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Merge :audio_processing and :aec_dump_interface.

Browse Source 
Merges the two targets in modules/audio_processing
and removes some redundant code. This enables not writing
a bunch of redundant code in
https://webrtc-review.googlesource.com/c/src/+/70502

':audio_processing' did depend on ':aec_dump_interface'.
'modules/audio_processing/aec_dump' did depend on
'aec_dump_interface' but not ':audio_processing'.

Having the AecDump implementation not depending on
'audio_processing' allows to have faster compilation time and
reduces the dependencies. However, maintaining such a decoupling
makes APM and AecDump client code more complex.

NOTRY=true # want this in and 'ios_api_framework' seems stuck.

Bug: webrtc:7404
Change-Id: I75a5f234591014ac42d52bc1a36526072f5be89c
Reviewed-on: https://webrtc-review.googlesource.com/76603
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23244}
This commit is contained in:
Alex Loiko
2018-05-15 13:13:22 +02:00
committed by Commit Bot
parent 28a325b523
commit 1aec594018
8 changed files with 27 additions and 71 deletions
Download Patch File Download Diff File Expand all files Collapse all files

18
modules/audio_processing/BUILD.gn
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@ -75,6 +75,8 @@ rtc_static_library("audio_processing") {
"gain_control_impl.h",
"gain_controller2.cc",
"gain_controller2.h",
"include/aec_dump.cc",
"include/aec_dump.h",
"include/audio_processing.cc",
"include/audio_processing.h",
"include/config.cc",
@ -116,7 +118,6 @@ rtc_static_library("audio_processing") {
defines = []
deps = [
":aec_core",
":aec_dump_interface",
":apm_logging",
":audio_frame_view",
":audio_generator_interface",
@ -199,20 +200,6 @@ rtc_source_set("audio_frame_view") {
]
}
rtc_source_set("aec_dump_interface") {
sources = [
"include/aec_dump.cc",
"include/aec_dump.h",
]
deps = [
":audio_frame_view",
"../../api:array_view",
"../../api/audio:audio_frame_api",
"../../rtc_base:rtc_base_approved",
]
}
rtc_source_set("audio_generator_interface") {
visibility = [ "*" ]
sources = [
@ -445,7 +432,6 @@ if (rtc_include_tests) {
"include/mock_audio_processing.h",
]
deps = [
":aec_dump_interface",
":audio_processing",
":audio_processing_statistics",
"../../test:test_support",

12
modules/audio_processing/aec_dump/BUILD.gn
View File

@ -15,7 +15,7 @@ rtc_source_set("aec_dump") {
]
deps = [
"..:aec_dump_interface",
"../",
"../../../rtc_base:rtc_base_approved",
]
}
@ -28,7 +28,7 @@ rtc_source_set("mock_aec_dump") {
]
deps = [
"..:aec_dump_interface",
"../",
"../../../test:test_support",
]
}
@ -42,7 +42,7 @@ rtc_source_set("mock_aec_dump_unittests") {
deps = [
":mock_aec_dump",
"..:audio_processing",
"../",
"../../../rtc_base:rtc_base_approved",
"//testing/gtest",
]
@ -61,7 +61,7 @@ if (rtc_enable_protobuf) {
deps = [
":aec_dump",
"..:aec_dump_interface",
"../",
"../../../api/audio:audio_frame_api",
"../../../rtc_base:checks",
"../../../rtc_base:protobuf_utils",
@ -80,8 +80,8 @@ if (rtc_enable_protobuf) {
deps = [
":aec_dump",
":aec_dump_impl",
"..:aec_dump_interface",
"..:audioproc_debug_proto",
"../",
"../../../rtc_base:rtc_task_queue",
"../../../test:fileutils",
"../../../test:test_support",
@ -101,6 +101,6 @@ rtc_source_set("null_aec_dump_factory") {
deps = [
":aec_dump",
"..:aec_dump_interface",
"../",
]
}

21
modules/audio_processing/aec_dump/aec_dump_impl.cc
View File

@ -74,26 +74,27 @@ AecDumpImpl::~AecDumpImpl() {
thread_sync_event.Wait(rtc::Event::kForever);
}
void AecDumpImpl::WriteInitMessage(
const InternalAPMStreamsConfig& streams_config) {
void AecDumpImpl::WriteInitMessage(const ProcessingConfig& api_format) {
auto task = CreateWriteToFileTask();
auto* event = task->GetEvent();
event->set_type(audioproc::Event::INIT);
audioproc::Init* msg = event->mutable_init();
msg->set_sample_rate(streams_config.input_sample_rate);
msg->set_output_sample_rate(streams_config.output_sample_rate);
msg->set_reverse_sample_rate(streams_config.render_input_sample_rate);
msg->set_reverse_output_sample_rate(streams_config.render_output_sample_rate);
msg->set_sample_rate(api_format.input_stream().sample_rate_hz());
msg->set_output_sample_rate(api_format.output_stream().sample_rate_hz());
msg->set_reverse_sample_rate(
api_format.reverse_input_stream().sample_rate_hz());
msg->set_reverse_output_sample_rate(
api_format.reverse_output_stream().sample_rate_hz());
msg->set_num_input_channels(
static_cast<int32_t>(streams_config.input_num_channels));
static_cast<int32_t>(api_format.input_stream().num_channels()));
msg->set_num_output_channels(
static_cast<int32_t>(streams_config.output_num_channels));
static_cast<int32_t>(api_format.output_stream().num_channels()));
msg->set_num_reverse_channels(
static_cast<int32_t>(streams_config.render_input_num_channels));
static_cast<int32_t>(api_format.reverse_input_stream().num_channels()));
msg->set_num_reverse_output_channels(
streams_config.render_output_num_channels);
api_format.reverse_output_stream().num_channels());
worker_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>(std::move(task)));
}

2
modules/audio_processing/aec_dump/aec_dump_impl.h
View File

@ -52,7 +52,7 @@ class AecDumpImpl : public AecDump {
~AecDumpImpl() override;
void WriteInitMessage(const InternalAPMStreamsConfig& api_format) override;
void WriteInitMessage(const ProcessingConfig& api_format) override;
void AddCaptureStreamInput(const AudioFrameView<const float>& src) override;
void AddCaptureStreamOutput(const AudioFrameView<const float>& src) override;

4
modules/audio_processing/aec_dump/aec_dump_unittest.cc
View File

@ -39,8 +39,8 @@ TEST(AecDumper, APICallsDoNotCrash) {
webrtc::InternalAPMConfig apm_config;
aec_dump->WriteConfig(apm_config);
webrtc::InternalAPMStreamsConfig streams_config;
aec_dump->WriteInitMessage(streams_config);
webrtc::ProcessingConfig api_format;
aec_dump->WriteInitMessage(api_format);
}
// Remove file after the AecDump d-tor has finished.
ASSERT_EQ(0, remove(filename.c_str()));

3
modules/audio_processing/aec_dump/mock_aec_dump.h
View File

@ -25,8 +25,7 @@ class MockAecDump : public AecDump {
MockAecDump();
virtual ~MockAecDump();
MOCK_METHOD1(WriteInitMessage,
void(const InternalAPMStreamsConfig& streams_config));
MOCK_METHOD1(WriteInitMessage, void(const ProcessingConfig& api_format));
MOCK_METHOD1(AddCaptureStreamInput,
void(const AudioFrameView<const float>& src));

22
modules/audio_processing/audio_processing_impl.cc
View File

@ -148,24 +148,6 @@ class HighPassFilterImpl : public HighPassFilter {
AudioProcessingImpl* apm_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(HighPassFilterImpl);
};
webrtc::InternalAPMStreamsConfig ToStreamsConfig(
const ProcessingConfig& api_format) {
webrtc::InternalAPMStreamsConfig result;
result.input_sample_rate = api_format.input_stream().sample_rate_hz();
result.input_num_channels = api_format.input_stream().num_channels();
result.output_num_channels = api_format.output_stream().num_channels();
result.render_input_num_channels =
api_format.reverse_input_stream().num_channels();
result.render_input_sample_rate =
api_format.reverse_input_stream().sample_rate_hz();
result.output_sample_rate = api_format.output_stream().sample_rate_hz();
result.render_output_sample_rate =
api_format.reverse_output_stream().sample_rate_hz();
result.render_output_num_channels =
api_format.reverse_output_stream().num_channels();
return result;
}
} // namespace
// Throughout webrtc, it's assumed that success is represented by zero.
@ -608,7 +590,7 @@ int AudioProcessingImpl::InitializeLocked() {
InitializePreProcessor();
if (aec_dump_) {
aec_dump_->WriteInitMessage(ToStreamsConfig(formats_.api_format));
aec_dump_->WriteInitMessage(formats_.api_format);
}
return kNoError;
}
@ -1638,7 +1620,7 @@ void AudioProcessingImpl::AttachAecDump(std::unique_ptr<AecDump> aec_dump) {
// 'aec_dump' parameter, which is after locks are released.
aec_dump_.swap(aec_dump);
WriteAecDumpConfigMessage(true);
aec_dump_->WriteInitMessage(ToStreamsConfig(formats_.api_format));
aec_dump_->WriteInitMessage(formats_.api_format);
}
void AudioProcessingImpl::DetachAecDump() {

16
modules/audio_processing/include/aec_dump.h
View File

@ -18,6 +18,7 @@
#include "api/array_view.h"
#include "api/audio/audio_frame.h"
#include "modules/audio_processing/include/audio_frame_view.h"
#include "modules/audio_processing/include/audio_processing.h"
namespace webrtc {
@ -55,18 +56,6 @@ struct InternalAPMConfig {
std::string experiments_description = "";
};
struct InternalAPMStreamsConfig {
int input_sample_rate = 0;
int output_sample_rate = 0;
int render_input_sample_rate = 0;
int render_output_sample_rate = 0;
size_t input_num_channels = 0;
size_t output_num_channels = 0;
size_t render_input_num_channels = 0;
size_t render_output_num_channels = 0;
};
// An interface for recording configuration and input/output streams
// of the Audio Processing Module. The recordings are called
// 'aec-dumps' and are stored in a protobuf format defined in
@ -87,8 +76,7 @@ class AecDump {
virtual ~AecDump() = default;
// Logs Event::Type INIT message.
virtual void WriteInitMessage(
const InternalAPMStreamsConfig& streams_config) = 0;
virtual void WriteInitMessage(const ProcessingConfig& api_format) = 0;
// Logs Event::Type STREAM message. To log an input/output pair,
// call the AddCapture* and AddAudioProcessingState methods followed