Remove RentACodec::GetEncoderStack
Callers can just remember the return value of
RentACodec::RentEncoderStack instead.
BUG=webrtc:5028
Review URL: https://codereview.webrtc.org/1612713002
Cr-Original-Commit-Position: refs/heads/master@{#11340}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 32be07bc36813d351c3555c502a819d3dd172d07
diff --git a/modules/audio_coding/acm2/audio_coding_module_impl.cc b/modules/audio_coding/acm2/audio_coding_module_impl.cc
index d0e02ea..af0b985 100644
--- a/modules/audio_coding/acm2/audio_coding_module_impl.cc
+++ b/modules/audio_coding/acm2/audio_coding_module_impl.cc
@@ -130,7 +130,6 @@
if (!HaveValidEncoder("Process"))
return -1;
- AudioEncoder* audio_encoder = rent_a_codec_.GetEncoderStack();
// Scale the timestamp to the codec's RTP timestamp rate.
uint32_t rtp_timestamp =
first_frame_ ? input_data.input_timestamp
@@ -138,20 +137,20 @@
rtc::CheckedDivExact(
input_data.input_timestamp - last_timestamp_,
static_cast<uint32_t>(rtc::CheckedDivExact(
- audio_encoder->SampleRateHz(),
- audio_encoder->RtpTimestampRateHz())));
+ encoder_stack_->SampleRateHz(),
+ encoder_stack_->RtpTimestampRateHz())));
last_timestamp_ = input_data.input_timestamp;
last_rtp_timestamp_ = rtp_timestamp;
first_frame_ = false;
- encode_buffer_.SetSize(audio_encoder->MaxEncodedBytes());
- encoded_info = audio_encoder->Encode(
+ encode_buffer_.SetSize(encoder_stack_->MaxEncodedBytes());
+ encoded_info = encoder_stack_->Encode(
rtp_timestamp, rtc::ArrayView<const int16_t>(
input_data.audio, input_data.audio_channel *
input_data.length_per_channel),
encode_buffer_.size(), encode_buffer_.data());
encode_buffer_.SetSize(encoded_info.encoded_bytes);
- bitrate_logger_.MaybeLog(audio_encoder->GetTargetBitrate() / 1000);
+ bitrate_logger_.MaybeLog(encoder_stack_->GetTargetBitrate() / 1000);
if (encode_buffer_.size() == 0 && !encoded_info.send_even_if_empty) {
// Not enough data.
return 0;
@@ -208,7 +207,7 @@
sp->speech_encoder = enc;
}
if (sp->speech_encoder)
- rent_a_codec_.RentEncoderStack(sp);
+ encoder_stack_ = rent_a_codec_.RentEncoderStack(sp);
return 0;
}
@@ -217,7 +216,7 @@
rtc::CritScope lock(&acm_crit_sect_);
auto* sp = codec_manager_.GetStackParams();
sp->speech_encoder = external_speech_encoder;
- rent_a_codec_.RentEncoderStack(sp);
+ encoder_stack_ = rent_a_codec_.RentEncoderStack(sp);
}
// Get current send codec.
@@ -240,21 +239,19 @@
"SendFrequency()");
rtc::CritScope lock(&acm_crit_sect_);
- const auto* enc = rent_a_codec_.GetEncoderStack();
- if (!enc) {
+ if (!encoder_stack_) {
WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, id_,
"SendFrequency Failed, no codec is registered");
return -1;
}
- return enc->SampleRateHz();
+ return encoder_stack_->SampleRateHz();
}
void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) {
rtc::CritScope lock(&acm_crit_sect_);
- auto* enc = rent_a_codec_.GetEncoderStack();
- if (enc) {
- enc->SetTargetBitrate(bitrate_bps);
+ if (encoder_stack_) {
+ encoder_stack_->SetTargetBitrate(bitrate_bps);
}
}
@@ -321,8 +318,7 @@
}
// Check whether we need an up-mix or down-mix?
- const size_t current_num_channels =
- rent_a_codec_.GetEncoderStack()->NumChannels();
+ const size_t current_num_channels = encoder_stack_->NumChannels();
const bool same_num_channels =
ptr_frame->num_channels_ == current_num_channels;
@@ -359,14 +355,15 @@
// is required, |*ptr_out| points to |in_frame|.
int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
const AudioFrame** ptr_out) {
- const auto* enc = rent_a_codec_.GetEncoderStack();
- const bool resample = in_frame.sample_rate_hz_ != enc->SampleRateHz();
+ const bool resample =
+ in_frame.sample_rate_hz_ != encoder_stack_->SampleRateHz();
// This variable is true if primary codec and secondary codec (if exists)
// are both mono and input is stereo.
// TODO(henrik.lundin): This condition should probably be
- // in_frame.num_channels_ > enc->NumChannels()
- const bool down_mix = in_frame.num_channels_ == 2 && enc->NumChannels() == 1;
+ // in_frame.num_channels_ > encoder_stack_->NumChannels()
+ const bool down_mix =
+ in_frame.num_channels_ == 2 && encoder_stack_->NumChannels() == 1;
if (!first_10ms_data_) {
expected_in_ts_ = in_frame.timestamp_;
@@ -376,8 +373,9 @@
// TODO(turajs): Do we need a warning here.
expected_codec_ts_ +=
(in_frame.timestamp_ - expected_in_ts_) *
- static_cast<uint32_t>(static_cast<double>(enc->SampleRateHz()) /
- static_cast<double>(in_frame.sample_rate_hz_));
+ static_cast<uint32_t>(
+ static_cast<double>(encoder_stack_->SampleRateHz()) /
+ static_cast<double>(in_frame.sample_rate_hz_));
expected_in_ts_ = in_frame.timestamp_;
}
@@ -416,7 +414,7 @@
dest_ptr_audio = preprocess_frame_.data_;
int samples_per_channel = resampler_.Resample10Msec(
- src_ptr_audio, in_frame.sample_rate_hz_, enc->SampleRateHz(),
+ src_ptr_audio, in_frame.sample_rate_hz_, encoder_stack_->SampleRateHz(),
preprocess_frame_.num_channels_, AudioFrame::kMaxDataSizeSamples,
dest_ptr_audio);
@@ -427,7 +425,7 @@
}
preprocess_frame_.samples_per_channel_ =
static_cast<size_t>(samples_per_channel);
- preprocess_frame_.sample_rate_hz_ = enc->SampleRateHz();
+ preprocess_frame_.sample_rate_hz_ = encoder_stack_->SampleRateHz();
}
expected_codec_ts_ +=
@@ -455,7 +453,7 @@
}
auto* sp = codec_manager_.GetStackParams();
if (sp->speech_encoder)
- rent_a_codec_.RentEncoderStack(sp);
+ encoder_stack_ = rent_a_codec_.RentEncoderStack(sp);
return 0;
#else
WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, id_,
@@ -480,7 +478,7 @@
}
auto* sp = codec_manager_.GetStackParams();
if (sp->speech_encoder)
- rent_a_codec_.RentEncoderStack(sp);
+ encoder_stack_ = rent_a_codec_.RentEncoderStack(sp);
if (enable_codec_fec) {
return sp->use_codec_fec ? 0 : -1;
} else {
@@ -492,8 +490,7 @@
int AudioCodingModuleImpl::SetPacketLossRate(int loss_rate) {
rtc::CritScope lock(&acm_crit_sect_);
if (HaveValidEncoder("SetPacketLossRate")) {
- rent_a_codec_.GetEncoderStack()->SetProjectedPacketLossRate(loss_rate /
- 100.0);
+ encoder_stack_->SetProjectedPacketLossRate(loss_rate / 100.0);
}
return 0;
}
@@ -512,7 +509,7 @@
}
auto* sp = codec_manager_.GetStackParams();
if (sp->speech_encoder)
- rent_a_codec_.RentEncoderStack(sp);
+ encoder_stack_ = rent_a_codec_.RentEncoderStack(sp);
return 0;
}
@@ -753,7 +750,7 @@
FATAL();
return 0;
}
- return rent_a_codec_.GetEncoderStack()->SetApplication(app) ? 0 : -1;
+ return encoder_stack_->SetApplication(app) ? 0 : -1;
}
// Informs Opus encoder of the maximum playback rate the receiver will render.
@@ -762,7 +759,7 @@
if (!HaveValidEncoder("SetOpusMaxPlaybackRate")) {
return -1;
}
- rent_a_codec_.GetEncoderStack()->SetMaxPlaybackRate(frequency_hz);
+ encoder_stack_->SetMaxPlaybackRate(frequency_hz);
return 0;
}
@@ -771,7 +768,7 @@
if (!HaveValidEncoder("EnableOpusDtx")) {
return -1;
}
- return rent_a_codec_.GetEncoderStack()->SetDtx(true) ? 0 : -1;
+ return encoder_stack_->SetDtx(true) ? 0 : -1;
}
int AudioCodingModuleImpl::DisableOpusDtx() {
@@ -779,7 +776,7 @@
if (!HaveValidEncoder("DisableOpusDtx")) {
return -1;
}
- return rent_a_codec_.GetEncoderStack()->SetDtx(false) ? 0 : -1;
+ return encoder_stack_->SetDtx(false) ? 0 : -1;
}
int AudioCodingModuleImpl::PlayoutTimestamp(uint32_t* timestamp) {
@@ -787,7 +784,7 @@
}
bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const {
- if (!rent_a_codec_.GetEncoderStack()) {
+ if (!encoder_stack_) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
"%s failed: No send codec is registered.", caller_name);
return false;
diff --git a/modules/audio_coding/acm2/audio_coding_module_impl.h b/modules/audio_coding/acm2/audio_coding_module_impl.h
index 19776d1..2a3bc61 100644
--- a/modules/audio_coding/acm2/audio_coding_module_impl.h
+++ b/modules/audio_coding/acm2/audio_coding_module_impl.h
@@ -251,6 +251,9 @@
CodecManager codec_manager_ GUARDED_BY(acm_crit_sect_);
RentACodec rent_a_codec_ GUARDED_BY(acm_crit_sect_);
+ // Last encoder stack obtained from rent_a_codec_.RentEncoderStack.
+ AudioEncoder* encoder_stack_ GUARDED_BY(acm_crit_sect_);
+
// This is to keep track of CN instances where we can send DTMFs.
uint8_t previous_pltype_ GUARDED_BY(acm_crit_sect_);
diff --git a/modules/audio_coding/acm2/rent_a_codec.cc b/modules/audio_coding/acm2/rent_a_codec.cc
index 5695fd6..44a38bb 100644
--- a/modules/audio_coding/acm2/rent_a_codec.cc
+++ b/modules/audio_coding/acm2/rent_a_codec.cc
@@ -280,21 +280,21 @@
// reset the latter to ensure its buffer is empty.
param->speech_encoder->Reset();
}
- encoder_stack_ = param->speech_encoder;
+ AudioEncoder* encoder_stack = param->speech_encoder;
if (param->use_red) {
- red_encoder_ = CreateRedEncoder(encoder_stack_, *red_pt);
+ red_encoder_ = CreateRedEncoder(encoder_stack, *red_pt);
if (red_encoder_)
- encoder_stack_ = red_encoder_.get();
+ encoder_stack = red_encoder_.get();
} else {
red_encoder_.reset();
}
if (param->use_cng) {
- cng_encoder_ = CreateCngEncoder(encoder_stack_, *cng_pt, param->vad_mode);
- encoder_stack_ = cng_encoder_.get();
+ cng_encoder_ = CreateCngEncoder(encoder_stack, *cng_pt, param->vad_mode);
+ encoder_stack = cng_encoder_.get();
} else {
cng_encoder_.reset();
}
- return encoder_stack_;
+ return encoder_stack;
}
AudioDecoder* RentACodec::RentIsacDecoder() {
diff --git a/modules/audio_coding/acm2/rent_a_codec.h b/modules/audio_coding/acm2/rent_a_codec.h
index b1dcc91..3f914ea 100644
--- a/modules/audio_coding/acm2/rent_a_codec.h
+++ b/modules/audio_coding/acm2/rent_a_codec.h
@@ -223,10 +223,6 @@
// the Rent-A-Codec is destroyed.
AudioEncoder* RentEncoderStack(StackParameters* param);
- // The last return value of RentEncoderStack, or null if it hasn't been
- // called.
- AudioEncoder* GetEncoderStack() const { return encoder_stack_; }
-
// Creates and returns an iSAC decoder, which will remain live until the
// Rent-A-Codec is destroyed. Subsequent calls will simply return the same
// object.
@@ -237,7 +233,6 @@
rtc::scoped_ptr<AudioEncoder> cng_encoder_;
rtc::scoped_ptr<AudioEncoder> red_encoder_;
rtc::scoped_ptr<AudioDecoder> isac_decoder_;
- AudioEncoder* encoder_stack_ = nullptr;
LockedIsacBandwidthInfo isac_bandwidth_info_;
RTC_DISALLOW_COPY_AND_ASSIGN(RentACodec);
diff --git a/modules/audio_coding/acm2/rent_a_codec_unittest.cc b/modules/audio_coding/acm2/rent_a_codec_unittest.cc
index e838488..435305d 100644
--- a/modules/audio_coding/acm2/rent_a_codec_unittest.cc
+++ b/modules/audio_coding/acm2/rent_a_codec_unittest.cc
@@ -135,7 +135,7 @@
EXPECT_CALL(external_encoder, Die());
}
- info = rac.GetEncoderStack()->Encode(0, audio, arraysize(encoded), encoded);
+ info = external_encoder.Encode(0, audio, arraysize(encoded), encoded);
EXPECT_EQ(0u, info.encoded_timestamp);
external_encoder.Mark("A");
@@ -147,13 +147,13 @@
EXPECT_EQ(param.speech_encoder, rac.RentEncoderStack(¶m));
// Don't expect any more calls to the external encoder.
- info = rac.GetEncoderStack()->Encode(1, audio, arraysize(encoded), encoded);
+ info = param.speech_encoder->Encode(1, audio, arraysize(encoded), encoded);
external_encoder.Mark("B");
// Change back to external encoder again.
param.speech_encoder = &external_encoder;
EXPECT_EQ(&external_encoder, rac.RentEncoderStack(¶m));
- info = rac.GetEncoderStack()->Encode(2, audio, arraysize(encoded), encoded);
+ info = external_encoder.Encode(2, audio, arraysize(encoded), encoded);
EXPECT_EQ(2u, info.encoded_timestamp);
}
