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Allow more than 2 input channels in AudioProcessing.

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The number of output channels is constrained to be equal to either 1 or the
number of input channels.

An earlier version of this commit caused a crash on AEC dump.

TBR=aluebs@webrtc.org,pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1248393003 .

Cr-Commit-Position: refs/heads/master@{#9626}
This commit is contained in:
Michael Graczyk
2015-07-23 11:41:39 -07:00
parent fcfdb08b59
commit 86c6d33aec
13 changed files with 711 additions and 374 deletions
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399
webrtc/modules/audio_processing/audio_processing_impl.cc
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@ -11,6 +11,7 @@
#include "webrtc/modules/audio_processing/audio_processing_impl.h"
#include <assert.h>
#include <algorithm>
#include "webrtc/base/checks.h"
#include "webrtc/base/platform_file.h"
@ -48,15 +49,32 @@ extern "C" {
#endif
#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
#define RETURN_ON_ERR(expr) \
do { \
int err = (expr); \
if (err != kNoError) { \
return err; \
} \
#define RETURN_ON_ERR(expr) \
do { \
int err = (expr); \
if (err != kNoError) { \
return err; \
} \
} while (0)
namespace webrtc {
namespace {
static bool LayoutHasKeyboard(AudioProcessing::ChannelLayout layout) {
switch (layout) {
case AudioProcessing::kMono:
case AudioProcessing::kStereo:
return false;
case AudioProcessing::kMonoAndKeyboard:
case AudioProcessing::kStereoAndKeyboard:
return true;
}
assert(false);
return false;
}
} // namespace
// Throughout webrtc, it's assumed that success is represented by zero.
static_assert(AudioProcessing::kNoError == 0, "kNoError must be zero");
@ -75,9 +93,7 @@ static_assert(AudioProcessing::kNoError == 0, "kNoError must be zero");
class GainControlForNewAgc : public GainControl, public VolumeCallbacks {
public:
explicit GainControlForNewAgc(GainControlImpl* gain_control)
: real_gain_control_(gain_control),
volume_(0) {
}
: real_gain_control_(gain_control), volume_(0) {}
// GainControl implementation.
int Enable(bool enable) override {
@ -166,10 +182,10 @@ AudioProcessingImpl::AudioProcessingImpl(const Config& config,
debug_file_(FileWrapper::Create()),
event_msg_(new audioproc::Event()),
#endif
fwd_in_format_(kSampleRate16kHz, 1),
api_format_({{{kSampleRate16kHz, 1, false},
{kSampleRate16kHz, 1, false},
{kSampleRate16kHz, 1, false}}}),
fwd_proc_format_(kSampleRate16kHz),
fwd_out_format_(kSampleRate16kHz, 1),
rev_in_format_(kSampleRate16kHz, 1),
rev_proc_format_(kSampleRate16kHz, 1),
split_rate_(kSampleRate16kHz),
stream_delay_ms_(0),
@ -253,12 +269,11 @@ int AudioProcessingImpl::Initialize() {
int AudioProcessingImpl::set_sample_rate_hz(int rate) {
CriticalSectionScoped crit_scoped(crit_);
return InitializeLocked(rate,
rate,
rev_in_format_.rate(),
fwd_in_format_.num_channels(),
fwd_out_format_.num_channels(),
rev_in_format_.num_channels());
ProcessingConfig processing_config = api_format_;
processing_config.input_stream().set_sample_rate_hz(rate);
processing_config.output_stream().set_sample_rate_hz(rate);
return InitializeLocked(processing_config);
}
int AudioProcessingImpl::Initialize(int input_sample_rate_hz,
@ -267,29 +282,39 @@ int AudioProcessingImpl::Initialize(int input_sample_rate_hz,
ChannelLayout input_layout,
ChannelLayout output_layout,
ChannelLayout reverse_layout) {
const ProcessingConfig processing_config = {
{{input_sample_rate_hz, ChannelsFromLayout(input_layout),
LayoutHasKeyboard(input_layout)},
{output_sample_rate_hz, ChannelsFromLayout(output_layout),
LayoutHasKeyboard(output_layout)},
{reverse_sample_rate_hz, ChannelsFromLayout(reverse_layout),
LayoutHasKeyboard(reverse_layout)}}};
return Initialize(processing_config);
}
int AudioProcessingImpl::Initialize(const ProcessingConfig& processing_config) {
CriticalSectionScoped crit_scoped(crit_);
return InitializeLocked(input_sample_rate_hz,
output_sample_rate_hz,
reverse_sample_rate_hz,
ChannelsFromLayout(input_layout),
ChannelsFromLayout(output_layout),
ChannelsFromLayout(reverse_layout));
return InitializeLocked(processing_config);
}
int AudioProcessingImpl::InitializeLocked() {
const int fwd_audio_buffer_channels = beamformer_enabled_ ?
fwd_in_format_.num_channels() :
fwd_out_format_.num_channels();
render_audio_.reset(new AudioBuffer(rev_in_format_.samples_per_channel(),
rev_in_format_.num_channels(),
rev_proc_format_.samples_per_channel(),
rev_proc_format_.num_channels(),
rev_proc_format_.samples_per_channel()));
capture_audio_.reset(new AudioBuffer(fwd_in_format_.samples_per_channel(),
fwd_in_format_.num_channels(),
fwd_proc_format_.samples_per_channel(),
fwd_audio_buffer_channels,
fwd_out_format_.samples_per_channel()));
const int fwd_audio_buffer_channels =
beamformer_enabled_ ? api_format_.input_stream().num_channels()
: api_format_.output_stream().num_channels();
if (api_format_.reverse_stream().num_channels() > 0) {
render_audio_.reset(new AudioBuffer(
api_format_.reverse_stream().num_frames(),
api_format_.reverse_stream().num_channels(),
rev_proc_format_.num_frames(), rev_proc_format_.num_channels(),
rev_proc_format_.num_frames()));
} else {
render_audio_.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()));
// Initialize all components.
for (auto item : component_list_) {
@ -317,38 +342,38 @@ int AudioProcessingImpl::InitializeLocked() {
return kNoError;
}
int AudioProcessingImpl::InitializeLocked(int input_sample_rate_hz,
int output_sample_rate_hz,
int reverse_sample_rate_hz,
int num_input_channels,
int num_output_channels,
int num_reverse_channels) {
if (input_sample_rate_hz <= 0 ||
output_sample_rate_hz <= 0 ||
reverse_sample_rate_hz <= 0) {
return kBadSampleRateError;
int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
for (const auto& stream : config.streams) {
if (stream.num_channels() < 0) {
return kBadNumberChannelsError;
}
if (stream.num_channels() > 0 && stream.sample_rate_hz() <= 0) {
return kBadSampleRateError;
}
}
if (num_output_channels > num_input_channels) {
return kBadNumberChannelsError;
}
// Only mono and stereo supported currently.
if (num_input_channels > 2 || num_input_channels < 1 ||
num_output_channels > 2 || num_output_channels < 1 ||
num_reverse_channels > 2 || num_reverse_channels < 1) {
return kBadNumberChannelsError;
}
if (beamformer_enabled_ &&
(static_cast<size_t>(num_input_channels) != array_geometry_.size() ||
num_output_channels > 1)) {
const int num_in_channels = config.input_stream().num_channels();
const int num_out_channels = config.output_stream().num_channels();
// Need at least one input channel.
// Need either one output channel or as many outputs as there are inputs.
if (num_in_channels == 0 ||
!(num_out_channels == 1 || num_out_channels == num_in_channels)) {
return kBadNumberChannelsError;
}
fwd_in_format_.set(input_sample_rate_hz, num_input_channels);
fwd_out_format_.set(output_sample_rate_hz, num_output_channels);
rev_in_format_.set(reverse_sample_rate_hz, num_reverse_channels);
if (beamformer_enabled_ &&
(static_cast<size_t>(num_in_channels) != array_geometry_.size() ||
num_out_channels > 1)) {
return kBadNumberChannelsError;
}
api_format_ = config;
// We process at the closest native rate >= min(input rate, output rate)...
int min_proc_rate = std::min(fwd_in_format_.rate(), fwd_out_format_.rate());
const int min_proc_rate =
std::min(api_format_.input_stream().sample_rate_hz(),
api_format_.output_stream().sample_rate_hz());
int fwd_proc_rate;
if (supports_48kHz_ && min_proc_rate > kSampleRate32kHz) {
fwd_proc_rate = kSampleRate48kHz;
@ -364,15 +389,15 @@ int AudioProcessingImpl::InitializeLocked(int input_sample_rate_hz,
fwd_proc_rate = kSampleRate16kHz;
}
fwd_proc_format_.set(fwd_proc_rate);
fwd_proc_format_ = StreamConfig(fwd_proc_rate);
// We normally process the reverse stream at 16 kHz. Unless...
int rev_proc_rate = kSampleRate16kHz;
if (fwd_proc_format_.rate() == kSampleRate8kHz) {
if (fwd_proc_format_.sample_rate_hz() == kSampleRate8kHz) {
// ...the forward stream is at 8 kHz.
rev_proc_rate = kSampleRate8kHz;
} else {
if (rev_in_format_.rate() == kSampleRate32kHz) {
if (api_format_.reverse_stream().sample_rate_hz() == kSampleRate32kHz) {
// ...or the input is at 32 kHz, in which case we use the splitting
// filter rather than the resampler.
rev_proc_rate = kSampleRate32kHz;
@ -381,13 +406,13 @@ int AudioProcessingImpl::InitializeLocked(int input_sample_rate_hz,
// Always downmix the reverse stream to mono for analysis. This has been
// demonstrated to work well for AEC in most practical scenarios.
rev_proc_format_.set(rev_proc_rate, 1);
rev_proc_format_ = StreamConfig(rev_proc_rate, 1);
if (fwd_proc_format_.rate() == kSampleRate32kHz ||
fwd_proc_format_.rate() == kSampleRate48kHz) {
if (fwd_proc_format_.sample_rate_hz() == kSampleRate32kHz ||
fwd_proc_format_.sample_rate_hz() == kSampleRate48kHz) {
split_rate_ = kSampleRate16kHz;
} else {
split_rate_ = fwd_proc_format_.rate();
split_rate_ = fwd_proc_format_.sample_rate_hz();
}
return InitializeLocked();
@ -395,26 +420,12 @@ int AudioProcessingImpl::InitializeLocked(int input_sample_rate_hz,
// Calls InitializeLocked() if any of the audio parameters have changed from
// their current values.
int AudioProcessingImpl::MaybeInitializeLocked(int input_sample_rate_hz,
int output_sample_rate_hz,
int reverse_sample_rate_hz,
int num_input_channels,
int num_output_channels,
int num_reverse_channels) {
if (input_sample_rate_hz == fwd_in_format_.rate() &&
output_sample_rate_hz == fwd_out_format_.rate() &&
reverse_sample_rate_hz == rev_in_format_.rate() &&
num_input_channels == fwd_in_format_.num_channels() &&
num_output_channels == fwd_out_format_.num_channels() &&
num_reverse_channels == rev_in_format_.num_channels()) {
int AudioProcessingImpl::MaybeInitializeLocked(
const ProcessingConfig& processing_config) {
if (processing_config == api_format_) {
return kNoError;
}
return InitializeLocked(input_sample_rate_hz,
output_sample_rate_hz,
reverse_sample_rate_hz,
num_input_channels,
num_output_channels,
num_reverse_channels);
return InitializeLocked(processing_config);
}
void AudioProcessingImpl::SetExtraOptions(const Config& config) {
@ -431,16 +442,16 @@ void AudioProcessingImpl::SetExtraOptions(const Config& config) {
int AudioProcessingImpl::input_sample_rate_hz() const {
CriticalSectionScoped crit_scoped(crit_);
return fwd_in_format_.rate();
return api_format_.input_stream().sample_rate_hz();
}
int AudioProcessingImpl::sample_rate_hz() const {
CriticalSectionScoped crit_scoped(crit_);
return fwd_in_format_.rate();
return api_format_.input_stream().sample_rate_hz();
}
int AudioProcessingImpl::proc_sample_rate_hz() const {
return fwd_proc_format_.rate();
return fwd_proc_format_.sample_rate_hz();
}
int AudioProcessingImpl::proc_split_sample_rate_hz() const {
@ -452,11 +463,11 @@ int AudioProcessingImpl::num_reverse_channels() const {
}
int AudioProcessingImpl::num_input_channels() const {
return fwd_in_format_.num_channels();
return api_format_.input_stream().num_channels();
}
int AudioProcessingImpl::num_output_channels() const {
return fwd_out_format_.num_channels();
return api_format_.output_stream().num_channels();
}
void AudioProcessingImpl::set_output_will_be_muted(bool muted) {
@ -479,44 +490,60 @@ int AudioProcessingImpl::ProcessStream(const float* const* src,
int output_sample_rate_hz,
ChannelLayout output_layout,
float* const* dest) {
StreamConfig input_stream = 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();
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));
if (samples_per_channel != input_stream.num_frames()) {
return kBadDataLengthError;
}
return ProcessStream(src, input_stream, output_stream, dest);
}
int AudioProcessingImpl::ProcessStream(const float* const* src,
const StreamConfig& input_config,
const StreamConfig& output_config,
float* const* dest) {
CriticalSectionScoped crit_scoped(crit_);
if (!src || !dest) {
return kNullPointerError;
}
RETURN_ON_ERR(MaybeInitializeLocked(input_sample_rate_hz,
output_sample_rate_hz,
rev_in_format_.rate(),
ChannelsFromLayout(input_layout),
ChannelsFromLayout(output_layout),
rev_in_format_.num_channels()));
if (samples_per_channel != fwd_in_format_.samples_per_channel()) {
return kBadDataLengthError;
}
ProcessingConfig processing_config = 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());
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_file_->Open()) {
event_msg_->set_type(audioproc::Event::STREAM);
audioproc::Stream* msg = event_msg_->mutable_stream();
const size_t channel_size =
sizeof(float) * fwd_in_format_.samples_per_channel();
for (int i = 0; i < fwd_in_format_.num_channels(); ++i)
sizeof(float) * api_format_.input_stream().num_frames();
for (int i = 0; i < api_format_.input_stream().num_channels(); ++i)
msg->add_input_channel(src[i], channel_size);
}
#endif
capture_audio_->CopyFrom(src, samples_per_channel, input_layout);
capture_audio_->CopyFrom(src, api_format_.input_stream());
RETURN_ON_ERR(ProcessStreamLocked());
capture_audio_->CopyTo(fwd_out_format_.samples_per_channel(),
output_layout,
dest);
capture_audio_->CopyTo(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) * fwd_out_format_.samples_per_channel();
for (int i = 0; i < fwd_out_format_.num_channels(); ++i)
sizeof(float) * api_format_.output_stream().num_frames();
for (int i = 0; i < api_format_.output_stream().num_channels(); ++i)
msg->add_output_channel(dest[i], channel_size);
RETURN_ON_ERR(WriteMessageToDebugFile());
}
@ -545,13 +572,14 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
// TODO(ajm): The input and output rates and channels are currently
// constrained to be identical in the int16 interface.
RETURN_ON_ERR(MaybeInitializeLocked(frame->sample_rate_hz_,
frame->sample_rate_hz_,
rev_in_format_.rate(),
frame->num_channels_,
frame->num_channels_,
rev_in_format_.num_channels()));
if (frame->samples_per_channel_ != fwd_in_format_.samples_per_channel()) {
ProcessingConfig processing_config = 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()) {
return kBadDataLengthError;
}
@ -559,9 +587,8 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
if (debug_file_->Open()) {
event_msg_->set_type(audioproc::Event::STREAM);
audioproc::Stream* msg = event_msg_->mutable_stream();
const size_t data_size = sizeof(int16_t) *
frame->samples_per_channel_ *
frame->num_channels_;
const size_t data_size =
sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
msg->set_input_data(frame->data_, data_size);
}
#endif
@ -573,9 +600,8 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_file_->Open()) {
audioproc::Stream* msg = event_msg_->mutable_stream();
const size_t data_size = sizeof(int16_t) *
frame->samples_per_channel_ *
frame->num_channels_;
const size_t data_size =
sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
msg->set_output_data(frame->data_, data_size);
RETURN_ON_ERR(WriteMessageToDebugFile());
}
@ -584,7 +610,6 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
return kNoError;
}
int AudioProcessingImpl::ProcessStreamLocked() {
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
if (debug_file_->Open()) {
@ -600,9 +625,8 @@ int AudioProcessingImpl::ProcessStreamLocked() {
AudioBuffer* ca = capture_audio_.get(); // For brevity.
if (use_new_agc_ && gain_control_->is_enabled()) {
agc_manager_->AnalyzePreProcess(ca->channels()[0],
ca->num_channels(),
fwd_proc_format_.samples_per_channel());
agc_manager_->AnalyzePreProcess(ca->channels()[0], ca->num_channels(),
fwd_proc_format_.num_frames());
}
bool data_processed = is_data_processed();
@ -627,12 +651,10 @@ int AudioProcessingImpl::ProcessStreamLocked() {
RETURN_ON_ERR(echo_control_mobile_->ProcessCaptureAudio(ca));
RETURN_ON_ERR(voice_detection_->ProcessCaptureAudio(ca));
if (use_new_agc_ &&
gain_control_->is_enabled() &&
if (use_new_agc_ && gain_control_->is_enabled() &&
(!beamformer_enabled_ || beamformer_->is_target_present())) {
agc_manager_->Process(ca->split_bands_const(0)[kBand0To8kHz],
ca->num_frames_per_band(),
split_rate_);
ca->num_frames_per_band(), split_rate_);
}
RETURN_ON_ERR(gain_control_->ProcessCaptureAudio(ca));
@ -646,15 +668,11 @@ int AudioProcessingImpl::ProcessStreamLocked() {
float voice_probability =
agc_manager_.get() ? agc_manager_->voice_probability() : 1.f;
transient_suppressor_->Suppress(ca->channels_f()[0],
ca->num_frames(),
ca->num_channels(),
ca->split_bands_const_f(0)[kBand0To8kHz],
ca->num_frames_per_band(),
ca->keyboard_data(),
ca->num_keyboard_frames(),
voice_probability,
key_pressed_);
transient_suppressor_->Suppress(
ca->channels_f()[0], ca->num_frames(), ca->num_channels(),
ca->split_bands_const_f(0)[kBand0To8kHz], ca->num_frames_per_band(),
ca->keyboard_data(), ca->num_keyboard_frames(), voice_probability,
key_pressed_);
}
// The level estimator operates on the recombined data.
@ -668,35 +686,47 @@ int AudioProcessingImpl::AnalyzeReverseStream(const float* const* data,
int samples_per_channel,
int sample_rate_hz,
ChannelLayout layout) {
const StreamConfig reverse_config = {
sample_rate_hz, ChannelsFromLayout(layout), LayoutHasKeyboard(layout),
};
if (samples_per_channel != reverse_config.num_frames()) {
return kBadDataLengthError;
}
return AnalyzeReverseStream(data, reverse_config);
}
int AudioProcessingImpl::AnalyzeReverseStream(
const float* const* data,
const StreamConfig& reverse_config) {
CriticalSectionScoped crit_scoped(crit_);
if (data == NULL) {
return kNullPointerError;
}
const int num_channels = ChannelsFromLayout(layout);
RETURN_ON_ERR(MaybeInitializeLocked(fwd_in_format_.rate(),
fwd_out_format_.rate(),
sample_rate_hz,
fwd_in_format_.num_channels(),
fwd_out_format_.num_channels(),
num_channels));
if (samples_per_channel != rev_in_format_.samples_per_channel()) {
return kBadDataLengthError;
if (reverse_config.num_channels() <= 0) {
return kBadNumberChannelsError;
}
ProcessingConfig processing_config = api_format_;
processing_config.reverse_stream() = reverse_config;
RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
assert(reverse_config.num_frames() ==
api_format_.reverse_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) * rev_in_format_.samples_per_channel();
for (int i = 0; i < num_channels; ++i)
sizeof(float) * api_format_.reverse_stream().num_frames();
for (int i = 0; i < api_format_.reverse_stream().num_channels(); ++i)
msg->add_channel(data[i], channel_size);
RETURN_ON_ERR(WriteMessageToDebugFile());
}
#endif
render_audio_->CopyFrom(data, samples_per_channel, layout);
render_audio_->CopyFrom(data, api_format_.reverse_stream());
return AnalyzeReverseStreamLocked();
}
@ -713,17 +743,21 @@ int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
return kBadSampleRateError;
}
// This interface does not tolerate different forward and reverse rates.
if (frame->sample_rate_hz_ != fwd_in_format_.rate()) {
if (frame->sample_rate_hz_ != api_format_.input_stream().sample_rate_hz()) {
return kBadSampleRateError;
}
RETURN_ON_ERR(MaybeInitializeLocked(fwd_in_format_.rate(),
fwd_out_format_.rate(),
frame->sample_rate_hz_,
fwd_in_format_.num_channels(),
fwd_in_format_.num_channels(),
frame->num_channels_));
if (frame->samples_per_channel_ != rev_in_format_.samples_per_channel()) {
if (frame->num_channels_ <= 0) {
return kBadNumberChannelsError;
}
ProcessingConfig processing_config = api_format_;
processing_config.reverse_stream().set_sample_rate_hz(frame->sample_rate_hz_);
processing_config.reverse_stream().set_num_channels(frame->num_channels_);
RETURN_ON_ERR(MaybeInitializeLocked(processing_config));
if (frame->samples_per_channel_ !=
api_format_.reverse_stream().num_frames()) {
return kBadDataLengthError;
}
@ -731,9 +765,8 @@ int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
if (debug_file_->Open()) {
event_msg_->set_type(audioproc::Event::REVERSE_STREAM);
audioproc::ReverseStream* msg = event_msg_->mutable_reverse_stream();
const size_t data_size = sizeof(int16_t) *
frame->samples_per_channel_ *
frame->num_channels_;
const size_t data_size =
sizeof(int16_t) * frame->samples_per_channel_ * frame->num_channels_;
msg->set_data(frame->data_, data_size);
RETURN_ON_ERR(WriteMessageToDebugFile());
}
@ -745,7 +778,7 @@ int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
int AudioProcessingImpl::AnalyzeReverseStreamLocked() {
AudioBuffer* ra = render_audio_.get(); // For brevity.
if (rev_proc_format_.rate() == kSampleRate32kHz) {
if (rev_proc_format_.sample_rate_hz() == kSampleRate32kHz) {
ra->SplitIntoFrequencyBands();
}
@ -947,13 +980,15 @@ bool AudioProcessingImpl::is_data_processed() const {
bool AudioProcessingImpl::output_copy_needed(bool is_data_processed) const {
// Check if we've upmixed or downmixed the audio.
return ((fwd_out_format_.num_channels() != fwd_in_format_.num_channels()) ||
return ((api_format_.output_stream().num_channels() !=
api_format_.input_stream().num_channels()) ||
is_data_processed || transient_suppressor_enabled_);
}
bool AudioProcessingImpl::synthesis_needed(bool is_data_processed) const {
return (is_data_processed && (fwd_proc_format_.rate() == kSampleRate32kHz ||
fwd_proc_format_.rate() == kSampleRate48kHz));
return (is_data_processed &&
(fwd_proc_format_.sample_rate_hz() == kSampleRate32kHz ||
fwd_proc_format_.sample_rate_hz() == kSampleRate48kHz));
}
bool AudioProcessingImpl::analysis_needed(bool is_data_processed) const {
@ -961,8 +996,8 @@ bool AudioProcessingImpl::analysis_needed(bool is_data_processed) const {
!transient_suppressor_enabled_) {
// Only level_estimator_ is enabled.
return false;
} else if (fwd_proc_format_.rate() == kSampleRate32kHz ||
fwd_proc_format_.rate() == kSampleRate48kHz) {
} else if (fwd_proc_format_.sample_rate_hz() == kSampleRate32kHz ||
fwd_proc_format_.sample_rate_hz() == kSampleRate48kHz) {
// Something besides level_estimator_ is enabled, and we have super-wb.
return true;
}
@ -986,9 +1021,9 @@ void AudioProcessingImpl::InitializeTransient() {
if (!transient_suppressor_.get()) {
transient_suppressor_.reset(new TransientSuppressor());
}
transient_suppressor_->Initialize(fwd_proc_format_.rate(),
split_rate_,
fwd_out_format_.num_channels());
transient_suppressor_->Initialize(
fwd_proc_format_.sample_rate_hz(), split_rate_,
api_format_.output_stream().num_channels());
}
}
@ -1031,8 +1066,8 @@ void AudioProcessingImpl::MaybeUpdateHistograms() {
const int frames_per_ms = rtc::CheckedDivExact(split_rate_, 1000);
const int aec_system_delay_ms =
WebRtcAec_system_delay(echo_cancellation()->aec_core()) / frames_per_ms;
const int diff_aec_system_delay_ms = aec_system_delay_ms -
last_aec_system_delay_ms_;
const int diff_aec_system_delay_ms =
aec_system_delay_ms - last_aec_system_delay_ms_;
if (diff_aec_system_delay_ms > kMinDiffDelayMs &&
last_aec_system_delay_ms_ != 0) {
RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AecSystemDelayJump",
@ -1072,8 +1107,8 @@ int AudioProcessingImpl::WriteMessageToDebugFile() {
return kUnspecifiedError;
}
#if defined(WEBRTC_ARCH_BIG_ENDIAN)
// TODO(ajm): Use little-endian "on the wire". For the moment, we can be
// pretty safe in assuming little-endian.
// TODO(ajm): Use little-endian "on the wire". For the moment, we can be
// pretty safe in assuming little-endian.
#endif
if (!event_msg_->SerializeToString(&event_str_)) {
@ -1096,12 +1131,12 @@ int AudioProcessingImpl::WriteMessageToDebugFile() {
int AudioProcessingImpl::WriteInitMessage() {
event_msg_->set_type(audioproc::Event::INIT);
audioproc::Init* msg = event_msg_->mutable_init();
msg->set_sample_rate(fwd_in_format_.rate());
msg->set_num_input_channels(fwd_in_format_.num_channels());
msg->set_num_output_channels(fwd_out_format_.num_channels());
msg->set_num_reverse_channels(rev_in_format_.num_channels());
msg->set_reverse_sample_rate(rev_in_format_.rate());
msg->set_output_sample_rate(fwd_out_format_.rate());
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_num_reverse_channels(api_format_.reverse_stream().num_channels());
msg->set_reverse_sample_rate(api_format_.reverse_stream().sample_rate_hz());
msg->set_output_sample_rate(api_format_.output_stream().sample_rate_hz());
int err = WriteMessageToDebugFile();
if (err != kNoError) {