Preparational work before introducing the locks in order to harmonize the code:
-Moved the initialize function
-Moved api_format into the shared state
BUG=
Review URL: https://codereview.webrtc.org/1413093002
Cr-Commit-Position: refs/heads/master@{#10668}
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index 3105224..f4977d4 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -195,10 +195,6 @@
debug_file_(FileWrapper::Create()),
event_msg_(new audioproc::Event()),
#endif
- api_format_({{{kSampleRate16kHz, 1, false},
- {kSampleRate16kHz, 1, false},
- {kSampleRate16kHz, 1, false},
- {kSampleRate16kHz, 1, false}}}),
fwd_proc_format_(kSampleRate16kHz),
rev_proc_format_(kSampleRate16kHz, 1),
split_rate_(kSampleRate16kHz),
@@ -310,26 +306,37 @@
return InitializeLocked(processing_config);
}
+// Calls InitializeLocked() if any of the audio parameters have changed from
+// their current values.
+int AudioProcessingImpl::MaybeInitializeLocked(
+ const ProcessingConfig& processing_config) {
+ if (processing_config == shared_state_.api_format_) {
+ return kNoError;
+ }
+ return InitializeLocked(processing_config);
+}
+
int AudioProcessingImpl::InitializeLocked() {
const int fwd_audio_buffer_channels =
- beamformer_enabled_ ? api_format_.input_stream().num_channels()
- : api_format_.output_stream().num_channels();
+ beamformer_enabled_
+ ? shared_state_.api_format_.input_stream().num_channels()
+ : shared_state_.api_format_.output_stream().num_channels();
const int rev_audio_buffer_out_num_frames =
- api_format_.reverse_output_stream().num_frames() == 0
+ shared_state_.api_format_.reverse_output_stream().num_frames() == 0
? rev_proc_format_.num_frames()
- : api_format_.reverse_output_stream().num_frames();
- if (api_format_.reverse_input_stream().num_channels() > 0) {
+ : shared_state_.api_format_.reverse_output_stream().num_frames();
+ if (shared_state_.api_format_.reverse_input_stream().num_channels() > 0) {
render_audio_.reset(new AudioBuffer(
- api_format_.reverse_input_stream().num_frames(),
- api_format_.reverse_input_stream().num_channels(),
+ shared_state_.api_format_.reverse_input_stream().num_frames(),
+ shared_state_.api_format_.reverse_input_stream().num_channels(),
rev_proc_format_.num_frames(), rev_proc_format_.num_channels(),
rev_audio_buffer_out_num_frames));
if (rev_conversion_needed()) {
render_converter_ = AudioConverter::Create(
- api_format_.reverse_input_stream().num_channels(),
- api_format_.reverse_input_stream().num_frames(),
- api_format_.reverse_output_stream().num_channels(),
- api_format_.reverse_output_stream().num_frames());
+ shared_state_.api_format_.reverse_input_stream().num_channels(),
+ shared_state_.api_format_.reverse_input_stream().num_frames(),
+ shared_state_.api_format_.reverse_output_stream().num_channels(),
+ shared_state_.api_format_.reverse_output_stream().num_frames());
} else {
render_converter_.reset(nullptr);
}
@@ -337,10 +344,11 @@
render_audio_.reset(nullptr);
render_converter_.reset(nullptr);
}
- capture_audio_.reset(new AudioBuffer(
- api_format_.input_stream().num_frames(),
- api_format_.input_stream().num_channels(), fwd_proc_format_.num_frames(),
- fwd_audio_buffer_channels, api_format_.output_stream().num_frames()));
+ capture_audio_.reset(
+ new AudioBuffer(shared_state_.api_format_.input_stream().num_frames(),
+ shared_state_.api_format_.input_stream().num_channels(),
+ fwd_proc_format_.num_frames(), fwd_audio_buffer_channels,
+ shared_state_.api_format_.output_stream().num_frames()));
// Initialize all components.
for (auto item : component_list_) {
@@ -396,12 +404,12 @@
return kBadNumberChannelsError;
}
- api_format_ = config;
+ shared_state_.api_format_ = config;
// We process at the closest native rate >= min(input rate, output rate)...
const int min_proc_rate =
- std::min(api_format_.input_stream().sample_rate_hz(),
- api_format_.output_stream().sample_rate_hz());
+ std::min(shared_state_.api_format_.input_stream().sample_rate_hz(),
+ shared_state_.api_format_.output_stream().sample_rate_hz());
int fwd_proc_rate;
for (size_t i = 0; i < kNumNativeSampleRates; ++i) {
fwd_proc_rate = kNativeSampleRatesHz[i];
@@ -423,7 +431,7 @@
// ...the forward stream is at 8 kHz.
rev_proc_rate = kSampleRate8kHz;
} else {
- if (api_format_.reverse_input_stream().sample_rate_hz() ==
+ if (shared_state_.api_format_.reverse_input_stream().sample_rate_hz() ==
kSampleRate32kHz) {
// ...or the input is at 32 kHz, in which case we use the splitting
// filter rather than the resampler.
@@ -445,15 +453,6 @@
return InitializeLocked();
}
-// Calls InitializeLocked() if any of the audio parameters have changed from
-// their current values.
-int AudioProcessingImpl::MaybeInitializeLocked(
- const ProcessingConfig& processing_config) {
- if (processing_config == api_format_) {
- return kNoError;
- }
- return InitializeLocked(processing_config);
-}
void AudioProcessingImpl::SetExtraOptions(const Config& config) {
CriticalSectionScoped crit_scoped(crit_);
@@ -481,11 +480,11 @@
}
int AudioProcessingImpl::num_input_channels() const {
- return api_format_.input_stream().num_channels();
+ return shared_state_.api_format_.input_stream().num_channels();
}
int AudioProcessingImpl::num_output_channels() const {
- return api_format_.output_stream().num_channels();
+ return shared_state_.api_format_.output_stream().num_channels();
}
void AudioProcessingImpl::set_output_will_be_muted(bool muted) {
@@ -505,12 +504,12 @@
ChannelLayout output_layout,
float* const* dest) {
CriticalSectionScoped crit_scoped(crit_);
- StreamConfig input_stream = api_format_.input_stream();
+ StreamConfig input_stream = shared_state_.api_format_.input_stream();
input_stream.set_sample_rate_hz(input_sample_rate_hz);
input_stream.set_num_channels(ChannelsFromLayout(input_layout));
input_stream.set_has_keyboard(LayoutHasKeyboard(input_layout));
- StreamConfig output_stream = api_format_.output_stream();
+ StreamConfig output_stream = shared_state_.api_format_.output_stream();
output_stream.set_sample_rate_hz(output_sample_rate_hz);
output_stream.set_num_channels(ChannelsFromLayout(output_layout));
output_stream.set_has_keyboard(LayoutHasKeyboard(output_layout));
@@ -534,13 +533,13 @@
echo_control_mobile_->ReadQueuedRenderData();
gain_control_->ReadQueuedRenderData();
- ProcessingConfig processing_config = api_format_;
+ ProcessingConfig processing_config = shared_state_.api_format_;
processing_config.input_stream() = input_config;
processing_config.output_stream() = output_config;
RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
assert(processing_config.input_stream().num_frames() ==
- api_format_.input_stream().num_frames());
+ shared_state_.api_format_.input_stream().num_frames());
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_file_->Open()) {
@@ -549,22 +548,24 @@
event_msg_->set_type(audioproc::Event::STREAM);
audioproc::Stream* msg = event_msg_->mutable_stream();
const size_t channel_size =
- sizeof(float) * api_format_.input_stream().num_frames();
- for (int i = 0; i < api_format_.input_stream().num_channels(); ++i)
+ sizeof(float) * shared_state_.api_format_.input_stream().num_frames();
+ for (int i = 0; i < shared_state_.api_format_.input_stream().num_channels();
+ ++i)
msg->add_input_channel(src[i], channel_size);
}
#endif
- capture_audio_->CopyFrom(src, api_format_.input_stream());
+ capture_audio_->CopyFrom(src, shared_state_.api_format_.input_stream());
RETURN_ON_ERR(ProcessStreamLocked());
- capture_audio_->CopyTo(api_format_.output_stream(), dest);
+ capture_audio_->CopyTo(shared_state_.api_format_.output_stream(), dest);
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_file_->Open()) {
audioproc::Stream* msg = event_msg_->mutable_stream();
const size_t channel_size =
- sizeof(float) * api_format_.output_stream().num_frames();
- for (int i = 0; i < api_format_.output_stream().num_channels(); ++i)
+ sizeof(float) * shared_state_.api_format_.output_stream().num_frames();
+ for (int i = 0;
+ i < shared_state_.api_format_.output_stream().num_channels(); ++i)
msg->add_output_channel(dest[i], channel_size);
RETURN_ON_ERR(WriteMessageToDebugFile());
}
@@ -589,6 +590,7 @@
frame->sample_rate_hz_ != kSampleRate48kHz) {
return kBadSampleRateError;
}
+
if (echo_control_mobile_->is_enabled() &&
frame->sample_rate_hz_ > kMaxAECMSampleRateHz) {
LOG(LS_ERROR) << "AECM only supports 16 or 8 kHz sample rates";
@@ -597,14 +599,15 @@
// TODO(ajm): The input and output rates and channels are currently
// constrained to be identical in the int16 interface.
- ProcessingConfig processing_config = api_format_;
+ ProcessingConfig processing_config = shared_state_.api_format_;
processing_config.input_stream().set_sample_rate_hz(frame->sample_rate_hz_);
processing_config.input_stream().set_num_channels(frame->num_channels_);
processing_config.output_stream().set_sample_rate_hz(frame->sample_rate_hz_);
processing_config.output_stream().set_num_channels(frame->num_channels_);
RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
- if (frame->samples_per_channel_ != api_format_.input_stream().num_frames()) {
+ if (frame->samples_per_channel_ !=
+ shared_state_.api_format_.input_stream().num_frames()) {
return kBadDataLengthError;
}
@@ -734,7 +737,8 @@
RETURN_ON_ERR(
AnalyzeReverseStream(src, reverse_input_config, reverse_output_config));
if (is_rev_processed()) {
- render_audio_->CopyTo(api_format_.reverse_output_stream(), dest);
+ render_audio_->CopyTo(shared_state_.api_format_.reverse_output_stream(),
+ dest);
} else if (rev_conversion_needed()) {
render_converter_->Convert(src, reverse_input_config.num_samples(), dest,
reverse_output_config.num_samples());
@@ -759,27 +763,31 @@
return kBadNumberChannelsError;
}
- ProcessingConfig processing_config = api_format_;
+ ProcessingConfig processing_config = shared_state_.api_format_;
processing_config.reverse_input_stream() = reverse_input_config;
processing_config.reverse_output_stream() = reverse_output_config;
RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
assert(reverse_input_config.num_frames() ==
- api_format_.reverse_input_stream().num_frames());
+ shared_state_.api_format_.reverse_input_stream().num_frames());
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_file_->Open()) {
event_msg_->set_type(audioproc::Event::REVERSE_STREAM);
audioproc::ReverseStream* msg = event_msg_->mutable_reverse_stream();
const size_t channel_size =
- sizeof(float) * api_format_.reverse_input_stream().num_frames();
- for (int i = 0; i < api_format_.reverse_input_stream().num_channels(); ++i)
+ sizeof(float) *
+ shared_state_.api_format_.reverse_input_stream().num_frames();
+ for (int i = 0;
+ i < shared_state_.api_format_.reverse_input_stream().num_channels();
+ ++i)
msg->add_channel(src[i], channel_size);
RETURN_ON_ERR(WriteMessageToDebugFile());
}
#endif
- render_audio_->CopyFrom(src, api_format_.reverse_input_stream());
+ render_audio_->CopyFrom(src,
+ shared_state_.api_format_.reverse_input_stream());
return ProcessReverseStreamLocked();
}
@@ -805,7 +813,8 @@
return kBadSampleRateError;
}
// This interface does not tolerate different forward and reverse rates.
- if (frame->sample_rate_hz_ != api_format_.input_stream().sample_rate_hz()) {
+ if (frame->sample_rate_hz_ !=
+ shared_state_.api_format_.input_stream().sample_rate_hz()) {
return kBadSampleRateError;
}
@@ -813,7 +822,7 @@
return kBadNumberChannelsError;
}
- ProcessingConfig processing_config = api_format_;
+ ProcessingConfig processing_config = shared_state_.api_format_;
processing_config.reverse_input_stream().set_sample_rate_hz(
frame->sample_rate_hz_);
processing_config.reverse_input_stream().set_num_channels(
@@ -825,7 +834,7 @@
RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
if (frame->samples_per_channel_ !=
- api_format_.reverse_input_stream().num_frames()) {
+ shared_state_.api_format_.reverse_input_stream().num_frames()) {
return kBadDataLengthError;
}
@@ -1049,8 +1058,8 @@
bool AudioProcessingImpl::output_copy_needed(bool is_data_processed) const {
// Check if we've upmixed or downmixed the audio.
- return ((api_format_.output_stream().num_channels() !=
- api_format_.input_stream().num_channels()) ||
+ return ((shared_state_.api_format_.output_stream().num_channels() !=
+ shared_state_.api_format_.input_stream().num_channels()) ||
is_data_processed || transient_suppressor_enabled_);
}
@@ -1078,8 +1087,8 @@
}
bool AudioProcessingImpl::rev_conversion_needed() const {
- return (api_format_.reverse_input_stream() !=
- api_format_.reverse_output_stream());
+ return (shared_state_.api_format_.reverse_input_stream() !=
+ shared_state_.api_format_.reverse_output_stream());
}
void AudioProcessingImpl::InitializeExperimentalAgc() {
@@ -1101,7 +1110,7 @@
}
transient_suppressor_->Initialize(
fwd_proc_format_.sample_rate_hz(), split_rate_,
- api_format_.output_stream().num_channels());
+ shared_state_.api_format_.output_stream().num_channels());
}
}
@@ -1220,14 +1229,18 @@
int AudioProcessingImpl::WriteInitMessage() {
event_msg_->set_type(audioproc::Event::INIT);
audioproc::Init* msg = event_msg_->mutable_init();
- msg->set_sample_rate(api_format_.input_stream().sample_rate_hz());
- msg->set_num_input_channels(api_format_.input_stream().num_channels());
- msg->set_num_output_channels(api_format_.output_stream().num_channels());
+ msg->set_sample_rate(
+ shared_state_.api_format_.input_stream().sample_rate_hz());
+ msg->set_num_input_channels(
+ shared_state_.api_format_.input_stream().num_channels());
+ msg->set_num_output_channels(
+ shared_state_.api_format_.output_stream().num_channels());
msg->set_num_reverse_channels(
- api_format_.reverse_input_stream().num_channels());
+ shared_state_.api_format_.reverse_input_stream().num_channels());
msg->set_reverse_sample_rate(
- api_format_.reverse_input_stream().sample_rate_hz());
- msg->set_output_sample_rate(api_format_.output_stream().sample_rate_hz());
+ shared_state_.api_format_.reverse_input_stream().sample_rate_hz());
+ msg->set_output_sample_rate(
+ shared_state_.api_format_.output_stream().sample_rate_hz());
// TODO(ekmeyerson): Add reverse output fields to event_msg_.
RETURN_ON_ERR(WriteMessageToDebugFile());
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
index 542886e..72dfbf4 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -176,8 +176,17 @@
std::string last_serialized_config_;
#endif
- // Format of processing streams at input/output call sites.
- ProcessingConfig api_format_;
+ // State that is written to while holding both the render and capture locks
+ // but can be read while holding only one of the locks.
+ struct SharedState {
+ SharedState()
+ : // Format of processing streams at input/output call sites.
+ api_format_({{{kSampleRate16kHz, 1, false},
+ {kSampleRate16kHz, 1, false},
+ {kSampleRate16kHz, 1, false},
+ {kSampleRate16kHz, 1, false}}}) {}
+ ProcessingConfig api_format_;
+ } shared_state_;
// Only the rate and samples fields of fwd_proc_format_ are used because the
// forward processing number of channels is mutable and is tracked by the
