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// Copyright 2016 The Chromium Authors
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#include "third_party/blink/renderer/modules/mediastream/local_media_stream_audio_source.h"
#include <utility>
#include "base/strings/stringprintf.h"
#include "base/strings/to_string.h"
#include "base/task/single_thread_task_runner.h"
#include "media/audio/audio_source_parameters.h"
#include "third_party/blink/public/mojom/mediastream/media_stream.mojom-blink.h"
#include "third_party/blink/public/platform/modules/webrtc/webrtc_logging.h"
#include "third_party/blink/public/platform/platform.h"
#include "third_party/blink/public/web/web_local_frame.h"
#include "third_party/blink/renderer/core/frame/local_frame.h"
namespace blink {
LocalMediaStreamAudioSource::LocalMediaStreamAudioSource(
LocalFrame* consumer_frame,
const MediaStreamDevice& device,
const int* requested_buffer_size,
bool disable_local_echo,
const MediaStreamAudioProcessingLayout& processing_layout,
ConstraintsRepeatingCallback started_callback,
scoped_refptr<base::SingleThreadTaskRunner> task_runner)
: MediaStreamAudioSource(std::move(task_runner),
true /* is_local_source */,
disable_local_echo),
processing_layout_(processing_layout),
consumer_frame_(consumer_frame),
started_callback_(std::move(started_callback)) {
CHECK(!processing_layout_.NeedWebrtcAudioProcessing());
DVLOG(1) << "LocalMediaStreamAudioSource::LocalMediaStreamAudioSource("
"device.input="
<< device.input.AsHumanReadableString()
<< " requested_buffer_size=" << requested_buffer_size
<< " enable_system_echo_cancellation="
<< base::ToString(processing_layout_.AecIsPlatformProvided()) << ")"
<< " system AEC available: "
<< (!!(device.input.effects() &
media::AudioParameters::ECHO_CANCELLER)
? "YES"
: "NO");
MediaStreamDevice device_to_request(device);
device_to_request.input.set_effects(processing_layout_.platform_effects());
SetDevice(device_to_request);
media::AudioParameters params = device_to_request.input;
int frames_per_buffer = params.frames_per_buffer();
if (requested_buffer_size) {
frames_per_buffer = *requested_buffer_size;
}
// If the device buffer size was not provided, use a default.
if (frames_per_buffer <= 0) {
frames_per_buffer = (params.sample_rate() * kFallbackAudioLatencyMs) / 1000;
}
params.set_format(media::AudioParameters::AUDIO_PCM_LOW_LATENCY);
params.set_frames_per_buffer(frames_per_buffer);
SetFormat(params);
}
LocalMediaStreamAudioSource::~LocalMediaStreamAudioSource() {
DVLOG(1) << "LocalMediaStreamAudioSource::~LocalMediaStreamAudioSource()";
EnsureSourceIsStopped();
}
bool LocalMediaStreamAudioSource::EnsureSourceIsStarted() {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
if (source_)
return true;
std::string str = base::StringPrintf(
"LocalMediaStreamAudioSource::EnsureSourceIsStarted."
" channel_layout=%d, sample_rate=%d, buffer_size=%d"
", session_id=%s, effects=%d. ",
device().input.channel_layout(), device().input.sample_rate(),
device().input.frames_per_buffer(),
device().session_id().ToString().c_str(), device().input.effects());
WebRtcLogMessage(str);
DVLOG(1) << str;
// Sanity-check that the consuming WebLocalFrame still exists.
// This is required by AudioDeviceFactory.
if (!consumer_frame_)
return false;
VLOG(1) << "Starting local audio input device (session_id="
<< device().session_id() << ") with audio parameters={"
<< GetAudioParameters().AsHumanReadableString() << "}.";
auto* web_frame =
static_cast<WebLocalFrame*>(WebFrame::FromCoreFrame(consumer_frame_));
source_ = Platform::Current()->NewAudioCapturerSource(
web_frame, media::AudioSourceParameters(device().session_id()));
source_->Initialize(GetAudioParameters(), this);
source_->Start();
return true;
}
void LocalMediaStreamAudioSource::EnsureSourceIsStopped() {
DCHECK_CALLED_ON_VALID_THREAD(thread_checker_);
if (!source_)
return;
source_->Stop();
source_ = nullptr;
VLOG(1) << "Stopped local audio input device (session_id="
<< device().session_id() << ") with audio parameters={"
<< GetAudioParameters().AsHumanReadableString() << "}.";
}
void LocalMediaStreamAudioSource::OnCaptureStarted() {
started_callback_.Run(this, mojom::MediaStreamRequestResult::OK, "");
}
void LocalMediaStreamAudioSource::Capture(
const media::AudioBus* audio_bus,
base::TimeTicks audio_capture_time,
const media::AudioGlitchInfo& glitch_info,
double volume) {
DCHECK(audio_bus);
DeliverDataToTracks(*audio_bus, audio_capture_time, glitch_info);
}
void LocalMediaStreamAudioSource::OnCaptureError(
media::AudioCapturerSource::ErrorCode code,
const std::string& why) {
WebRtcLogMessage(
base::StringPrintf("LocalMediaStreamAudioSource::OnCaptureError: %d, %s",
static_cast<int>(code), why.c_str()));
StopSourceOnError(code, why);
}
void LocalMediaStreamAudioSource::OnCaptureMuted(bool is_muted) {
SetMutedState(is_muted);
}
void LocalMediaStreamAudioSource::ChangeSourceImpl(
const MediaStreamDevice& new_device) {
WebRtcLogMessage(
"LocalMediaStreamAudioSource::ChangeSourceImpl(new_device = " +
new_device.id + ")");
EnsureSourceIsStopped();
SetDevice(new_device);
EnsureSourceIsStarted();
}
std::optional<blink::AudioProcessingProperties>
LocalMediaStreamAudioSource::GetAudioProcessingProperties() const {
return processing_layout_.properties();
}
} // namespace blink
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