| // Copyright 2021 The Chromium Authors |
| // Use of this source code is governed by a BSD-style license that can be |
| // found in the LICENSE file. |
| |
| #include "media/webrtc/audio_processor.h" |
| |
| #include <stddef.h> |
| #include <stdint.h> |
| |
| #include <algorithm> |
| #include <array> |
| #include <limits> |
| #include <memory> |
| #include <utility> |
| |
| #include "base/feature_list.h" |
| #include "base/functional/callback_helpers.h" |
| #include "base/logging.h" |
| #include "base/strings/stringprintf.h" |
| #include "base/task/thread_pool.h" |
| #include "base/trace_event/trace_event.h" |
| #include "build/build_config.h" |
| #include "build/chromecast_buildflags.h" |
| #include "build/chromeos_buildflags.h" |
| #include "media/base/audio_fifo.h" |
| #include "media/base/audio_parameters.h" |
| #include "media/base/audio_timestamp_helper.h" |
| #include "media/base/channel_layout.h" |
| #include "media/base/limits.h" |
| #include "media/webrtc/constants.h" |
| #include "media/webrtc/helpers.h" |
| #include "media/webrtc/webrtc_features.h" |
| #include "third_party/abseil-cpp/absl/types/optional.h" |
| #include "third_party/webrtc/modules/audio_processing/include/audio_processing.h" |
| #include "third_party/webrtc_overrides/task_queue_factory.h" |
| |
| namespace media { |
| namespace { |
| constexpr int kBuffersPerSecond = 100; // 10 ms per buffer. |
| |
| int GetCaptureBufferSize(bool need_webrtc_processing, |
| const AudioParameters device_format) { |
| #if BUILDFLAG(IS_ANDROID) && !BUILDFLAG(IS_CAST_ANDROID) |
| // TODO(henrika): Re-evaluate whether to use same logic as other platforms. |
| // https://crbug.com/638081 |
| // Note: This computation does not match 2x10 ms as defined for audio |
| // processing when rates are 50 modulo 100. 22050 Hz here gives buffer size |
| // (2*22050)/100 = 441 samples, while WebRTC processes in chunks of 22050/100 |
| // = 220 samples. This leads to unnecessary rebuffering. |
| return 2 * device_format.sample_rate() / 100; |
| #else |
| const int buffer_size_10_ms = device_format.sample_rate() / 100; |
| // If audio processing is turned on, require 10ms buffers to avoid |
| // rebuffering. |
| if (need_webrtc_processing) { |
| DCHECK_EQ(buffer_size_10_ms, webrtc::AudioProcessing::GetFrameSize( |
| device_format.sample_rate())); |
| return buffer_size_10_ms; |
| } |
| |
| // If WebRTC audio processing is not required and the native hardware buffer |
| // size was provided, use it. It can be harmful, in terms of CPU/power |
| // consumption, to use smaller buffer sizes than the native size. |
| // (https://crbug.com/362261). |
| if (int hardware_buffer_size = device_format.frames_per_buffer()) |
| return hardware_buffer_size; |
| |
| // If the buffer size is missing from the device parameters, provide 10ms as |
| // a fall-back. |
| return buffer_size_10_ms; |
| #endif |
| } |
| |
| bool ApmNeedsPlayoutReference(const webrtc::AudioProcessing* apm, |
| const AudioProcessingSettings& settings) { |
| if (!base::FeatureList::IsEnabled( |
| features::kWebRtcApmTellsIfPlayoutReferenceIsNeeded)) { |
| return settings.NeedPlayoutReference(); |
| } |
| if (!apm) { |
| // APM is not available; hence, observing the playout reference is not |
| // needed. |
| return false; |
| } |
| // TODO(crbug.com/1410129): Move the logic below into WebRTC APM since APM may |
| // use injected sub-modules the usage of which is not reflected in the APM |
| // config (e.g., render side processing). |
| const webrtc::AudioProcessing::Config config = apm->GetConfig(); |
| const bool aec = config.echo_canceller.enabled; |
| const bool legacy_agc = |
| config.gain_controller1.enabled && |
| !config.gain_controller1.analog_gain_controller.enabled; |
| return aec || legacy_agc; |
| } |
| } // namespace |
| |
| // Wraps AudioBus to provide access to the array of channel pointers, since this |
| // is the type webrtc::AudioProcessing deals in. The array is refreshed on every |
| // channel_ptrs() call, and will be valid until the underlying AudioBus pointers |
| // are changed, e.g. through calls to SetChannelData() or SwapChannels(). |
| class AudioProcessorCaptureBus { |
| public: |
| AudioProcessorCaptureBus(int channels, int frames) |
| : bus_(media::AudioBus::Create(channels, frames)), |
| channel_ptrs_(new float*[channels]) { |
| bus_->Zero(); |
| } |
| |
| media::AudioBus* bus() { return bus_.get(); } |
| |
| float* const* channel_ptrs() { |
| for (int i = 0; i < bus_->channels(); ++i) { |
| channel_ptrs_[i] = bus_->channel(i); |
| } |
| return channel_ptrs_.get(); |
| } |
| |
| private: |
| std::unique_ptr<media::AudioBus> bus_; |
| std::unique_ptr<float*[]> channel_ptrs_; |
| }; |
| |
| // Wraps AudioFifo to provide a cleaner interface to AudioProcessor. |
| // It avoids the FIFO when the source and destination frames match. If |
| // |source_channels| is larger than |destination_channels|, only the first |
| // |destination_channels| are kept from the source. |
| // Does not support concurrent access. |
| class AudioProcessorCaptureFifo { |
| public: |
| AudioProcessorCaptureFifo(int source_channels, |
| int destination_channels, |
| int source_frames, |
| int destination_frames, |
| int sample_rate) |
| : |
| #if DCHECK_IS_ON() |
| source_channels_(source_channels), |
| source_frames_(source_frames), |
| #endif |
| sample_rate_(sample_rate), |
| destination_( |
| std::make_unique<AudioProcessorCaptureBus>(destination_channels, |
| destination_frames)), |
| data_available_(false) { |
| DCHECK_GE(source_channels, destination_channels); |
| |
| if (source_channels > destination_channels) { |
| audio_source_intermediate_ = |
| media::AudioBus::CreateWrapper(destination_channels); |
| } |
| |
| if (source_frames != destination_frames) { |
| // Since we require every Push to be followed by as many Consumes as |
| // possible, twice the larger of the two is a (probably) loose upper bound |
| // on the FIFO size. |
| const int fifo_frames = 2 * std::max(source_frames, destination_frames); |
| fifo_ = |
| std::make_unique<media::AudioFifo>(destination_channels, fifo_frames); |
| } |
| } |
| |
| void Push(const media::AudioBus& source, base::TimeDelta audio_delay) { |
| #if DCHECK_IS_ON() |
| DCHECK_EQ(source.channels(), source_channels_); |
| DCHECK_EQ(source.frames(), source_frames_); |
| #endif |
| const media::AudioBus* source_to_push = &source; |
| |
| if (audio_source_intermediate_) { |
| for (int i = 0; i < destination_->bus()->channels(); ++i) { |
| audio_source_intermediate_->SetChannelData( |
| i, const_cast<float*>(source.channel(i))); |
| } |
| audio_source_intermediate_->set_frames(source.frames()); |
| source_to_push = audio_source_intermediate_.get(); |
| } |
| |
| if (fifo_) { |
| CHECK_LT(fifo_->frames(), destination_->bus()->frames()); |
| next_audio_delay_ = |
| audio_delay + fifo_->frames() * base::Seconds(1) / sample_rate_; |
| fifo_->Push(source_to_push); |
| } else { |
| CHECK(!data_available_); |
| source_to_push->CopyTo(destination_->bus()); |
| next_audio_delay_ = audio_delay; |
| data_available_ = true; |
| } |
| } |
| |
| // Returns true if there are destination_frames() of data available to be |
| // consumed, and otherwise false. |
| bool Consume(AudioProcessorCaptureBus** destination, |
| base::TimeDelta* audio_delay) { |
| if (fifo_) { |
| if (fifo_->frames() < destination_->bus()->frames()) |
| return false; |
| |
| fifo_->Consume(destination_->bus(), 0, destination_->bus()->frames()); |
| *audio_delay = next_audio_delay_; |
| next_audio_delay_ -= |
| destination_->bus()->frames() * base::Seconds(1) / sample_rate_; |
| } else { |
| if (!data_available_) |
| return false; |
| *audio_delay = next_audio_delay_; |
| // The data was already copied to |destination_| in this case. |
| data_available_ = false; |
| } |
| |
| *destination = destination_.get(); |
| return true; |
| } |
| |
| private: |
| #if DCHECK_IS_ON() |
| const int source_channels_; |
| const int source_frames_; |
| #endif |
| const int sample_rate_; |
| std::unique_ptr<media::AudioBus> audio_source_intermediate_; |
| std::unique_ptr<AudioProcessorCaptureBus> destination_; |
| std::unique_ptr<media::AudioFifo> fifo_; |
| |
| // When using |fifo_|, this is the audio delay of the first sample to be |
| // consumed next from the FIFO. When not using |fifo_|, this is the audio |
| // delay of the first sample in |destination_|. |
| base::TimeDelta next_audio_delay_; |
| |
| // True when |destination_| contains the data to be returned by the next call |
| // to Consume(). Only used when the FIFO is disabled. |
| bool data_available_; |
| }; |
| |
| // static |
| std::unique_ptr<AudioProcessor> AudioProcessor::Create( |
| DeliverProcessedAudioCallback deliver_processed_audio_callback, |
| LogCallback log_callback, |
| const AudioProcessingSettings& settings, |
| const media::AudioParameters& input_format, |
| const media::AudioParameters& output_format) { |
| log_callback.Run(base::StringPrintf( |
| "AudioProcessor::Create({multi_channel_capture_processing=%s})", |
| settings.multi_channel_capture_processing ? "true" : "false")); |
| |
| rtc::scoped_refptr<webrtc::AudioProcessing> webrtc_audio_processing = |
| media::CreateWebRtcAudioProcessingModule(settings); |
| |
| return std::make_unique<AudioProcessor>( |
| std::move(deliver_processed_audio_callback), std::move(log_callback), |
| input_format, output_format, std::move(webrtc_audio_processing), |
| settings.stereo_mirroring, |
| ApmNeedsPlayoutReference(webrtc_audio_processing.get(), settings)); |
| } |
| |
| AudioProcessor::AudioProcessor( |
| DeliverProcessedAudioCallback deliver_processed_audio_callback, |
| LogCallback log_callback, |
| const media::AudioParameters& input_format, |
| const media::AudioParameters& output_format, |
| rtc::scoped_refptr<webrtc::AudioProcessing> webrtc_audio_processing, |
| bool stereo_mirroring, |
| bool needs_playout_reference) |
| : webrtc_audio_processing_(webrtc_audio_processing), |
| stereo_mirroring_(stereo_mirroring), |
| needs_playout_reference_(needs_playout_reference), |
| log_callback_(std::move(log_callback)), |
| input_format_(input_format), |
| output_format_(output_format), |
| deliver_processed_audio_callback_( |
| std::move(deliver_processed_audio_callback)), |
| audio_delay_stats_reporter_(kBuffersPerSecond), |
| playout_fifo_( |
| // Unretained is safe, since the callback is always called |
| // synchronously within the AudioProcessor. |
| base::BindRepeating(&AudioProcessor::AnalyzePlayoutData, |
| base::Unretained(this))) { |
| DCHECK(deliver_processed_audio_callback_); |
| DCHECK(log_callback_); |
| |
| CHECK(input_format_.IsValid()); |
| CHECK(output_format_.IsValid()); |
| if (webrtc_audio_processing_) { |
| DCHECK_EQ( |
| webrtc::AudioProcessing::GetFrameSize(output_format_.sample_rate()), |
| output_format_.frames_per_buffer()); |
| } |
| if (input_format_.sample_rate() % 100 != 0 || |
| output_format_.sample_rate() % 100 != 0) { |
| // The WebRTC audio processing module may simulate clock drift on |
| // non-divisible sample rates. |
| SendLogMessage(base::StringPrintf( |
| "%s: WARNING: Sample rate not divisible by 100, processing is provided " |
| "on a best-effort basis. input rate=[%d], output rate=[%d]", |
| __func__, input_format_.sample_rate(), output_format_.sample_rate())); |
| } |
| SendLogMessage(base::StringPrintf( |
| "%s({input_format_=[%s], output_format_=[%s]})", __func__, |
| input_format_.AsHumanReadableString().c_str(), |
| output_format_.AsHumanReadableString().c_str())); |
| |
| // If audio processing is needed, rebuffer to APM frame size. If not, rebuffer |
| // to the requested output format. |
| const int fifo_output_frames_per_buffer = |
| webrtc_audio_processing_ |
| ? webrtc::AudioProcessing::GetFrameSize(input_format_.sample_rate()) |
| : output_format_.frames_per_buffer(); |
| SendLogMessage(base::StringPrintf( |
| "%s => (capture FIFO: fifo_output_frames_per_buffer=%d)", __func__, |
| fifo_output_frames_per_buffer)); |
| capture_fifo_ = std::make_unique<AudioProcessorCaptureFifo>( |
| input_format.channels(), input_format_.channels(), |
| input_format.frames_per_buffer(), fifo_output_frames_per_buffer, |
| input_format.sample_rate()); |
| |
| if (webrtc_audio_processing_) { |
| output_bus_ = std::make_unique<AudioProcessorCaptureBus>( |
| output_format_.channels(), output_format.frames_per_buffer()); |
| } |
| } |
| |
| AudioProcessor::~AudioProcessor() { |
| DCHECK_CALLED_ON_VALID_SEQUENCE(owning_sequence_); |
| OnStopDump(); |
| } |
| |
| void AudioProcessor::ProcessCapturedAudio(const media::AudioBus& audio_source, |
| base::TimeTicks audio_capture_time, |
| int num_preferred_channels, |
| double volume, |
| bool key_pressed) { |
| DCHECK(deliver_processed_audio_callback_); |
| // Sanity-check the input audio format in debug builds. |
| DCHECK(input_format_.IsValid()); |
| DCHECK_EQ(audio_source.channels(), input_format_.channels()); |
| DCHECK_EQ(audio_source.frames(), input_format_.frames_per_buffer()); |
| |
| base::TimeDelta capture_delay = base::TimeTicks::Now() - audio_capture_time; |
| TRACE_EVENT1("audio", "AudioProcessor::ProcessCapturedAudio", "delay (ms)", |
| capture_delay.InMillisecondsF()); |
| |
| capture_fifo_->Push(audio_source, capture_delay); |
| |
| // Process and consume the data in the FIFO until there is not enough |
| // data to process. |
| AudioProcessorCaptureBus* process_bus; |
| while (capture_fifo_->Consume(&process_bus, &capture_delay)) { |
| // Use the process bus directly if audio processing is disabled. |
| AudioProcessorCaptureBus* output_bus = process_bus; |
| absl::optional<double> new_volume; |
| if (webrtc_audio_processing_) { |
| output_bus = output_bus_.get(); |
| new_volume = |
| ProcessData(process_bus->channel_ptrs(), process_bus->bus()->frames(), |
| capture_delay, volume, key_pressed, |
| num_preferred_channels, output_bus->channel_ptrs()); |
| } |
| |
| // Swap channels before interleaving the data. |
| if (stereo_mirroring_ && |
| output_format_.channel_layout() == media::CHANNEL_LAYOUT_STEREO) { |
| // Swap the first and second channels. |
| output_bus->bus()->SwapChannels(0, 1); |
| } |
| |
| deliver_processed_audio_callback_.Run(*output_bus->bus(), |
| audio_capture_time, new_volume); |
| } |
| } |
| |
| void AudioProcessor::SetOutputWillBeMuted(bool muted) { |
| DCHECK_CALLED_ON_VALID_SEQUENCE(owning_sequence_); |
| SendLogMessage( |
| base::StringPrintf("%s({muted=%s})", __func__, muted ? "true" : "false")); |
| if (webrtc_audio_processing_) { |
| webrtc_audio_processing_->set_output_will_be_muted(muted); |
| } |
| } |
| |
| void AudioProcessor::OnStartDump(base::File dump_file) { |
| DCHECK_CALLED_ON_VALID_SEQUENCE(owning_sequence_); |
| DCHECK(dump_file.IsValid()); |
| |
| if (webrtc_audio_processing_) { |
| if (!worker_queue_) { |
| worker_queue_ = std::make_unique<rtc::TaskQueue>( |
| CreateWebRtcTaskQueue(rtc::TaskQueue::Priority::LOW)); |
| } |
| // Here tasks will be posted on the |worker_queue_|. It must be |
| // kept alive until media::StopEchoCancellationDump is called or the |
| // webrtc::AudioProcessing instance is destroyed. |
| media::StartEchoCancellationDump(webrtc_audio_processing_.get(), |
| std::move(dump_file), worker_queue_.get()); |
| } else { |
| // Post the file close to avoid blocking the control sequence. |
| base::ThreadPool::PostTask( |
| FROM_HERE, {base::TaskPriority::LOWEST, base::MayBlock()}, |
| base::DoNothingWithBoundArgs(std::move(dump_file))); |
| } |
| } |
| |
| void AudioProcessor::OnStopDump() { |
| DCHECK_CALLED_ON_VALID_SEQUENCE(owning_sequence_); |
| if (!worker_queue_) |
| return; |
| if (webrtc_audio_processing_) |
| media::StopEchoCancellationDump(webrtc_audio_processing_.get()); |
| worker_queue_.reset(nullptr); |
| } |
| |
| void AudioProcessor::OnPlayoutData(const AudioBus& audio_bus, |
| int sample_rate, |
| base::TimeDelta audio_delay) { |
| TRACE_EVENT1("audio", "AudioProcessor::OnPlayoutData", "delay (ms)", |
| audio_delay.InMillisecondsF()); |
| |
| if (!webrtc_audio_processing_) { |
| return; |
| } |
| |
| unbuffered_playout_delay_ = audio_delay; |
| |
| if (!playout_sample_rate_hz_ || sample_rate != *playout_sample_rate_hz_) { |
| // We reset the buffer on sample rate changes because the current buffer |
| // content is rendered obsolete (the audio processing module will reset |
| // internally) and the FIFO does not resample previous content to the new |
| // rate. |
| // Channel count changes are already handled within the AudioPushFifo. |
| playout_sample_rate_hz_ = sample_rate; |
| const int samples_per_channel = |
| webrtc::AudioProcessing::GetFrameSize(sample_rate); |
| playout_fifo_.Reset(samples_per_channel); |
| } |
| |
| playout_fifo_.Push(audio_bus); |
| } |
| |
| void AudioProcessor::AnalyzePlayoutData(const AudioBus& audio_bus, |
| int frame_delay) { |
| DCHECK(webrtc_audio_processing_); |
| DCHECK(playout_sample_rate_hz_.has_value()); |
| |
| const base::TimeDelta playout_delay = |
| unbuffered_playout_delay_ + |
| AudioTimestampHelper::FramesToTime(frame_delay, *playout_sample_rate_hz_); |
| playout_delay_ = playout_delay; |
| TRACE_EVENT1("audio", "AudioProcessor::AnalyzePlayoutData", "delay (ms)", |
| playout_delay.InMillisecondsF()); |
| |
| webrtc::StreamConfig input_stream_config(*playout_sample_rate_hz_, |
| audio_bus.channels()); |
| std::array<const float*, media::limits::kMaxChannels> input_ptrs; |
| for (int i = 0; i < audio_bus.channels(); ++i) |
| input_ptrs[i] = audio_bus.channel(i); |
| |
| const int apm_error = webrtc_audio_processing_->AnalyzeReverseStream( |
| input_ptrs.data(), input_stream_config); |
| if (apm_error != webrtc::AudioProcessing::kNoError && |
| apm_playout_error_code_log_count_ < 10) { |
| LOG(ERROR) << "MSAP::OnPlayoutData: AnalyzeReverseStream error=" |
| << apm_error; |
| ++apm_playout_error_code_log_count_; |
| } |
| } |
| |
| webrtc::AudioProcessingStats AudioProcessor::GetStats() { |
| if (!webrtc_audio_processing_) |
| return {}; |
| return webrtc_audio_processing_->GetStatistics(); |
| } |
| |
| absl::optional<double> AudioProcessor::ProcessData( |
| const float* const* process_ptrs, |
| int process_frames, |
| base::TimeDelta capture_delay, |
| double volume, |
| bool key_pressed, |
| int num_preferred_channels, |
| float* const* output_ptrs) { |
| DCHECK(webrtc_audio_processing_); |
| |
| const base::TimeDelta playout_delay = playout_delay_; |
| |
| TRACE_EVENT2("audio", "AudioProcessor::ProcessData", "capture_delay (ms)", |
| capture_delay.InMillisecondsF(), "playout_delay (ms)", |
| playout_delay.InMillisecondsF()); |
| |
| const int64_t total_delay_ms = |
| (capture_delay + playout_delay).InMilliseconds(); |
| |
| if (total_delay_ms > 300 && large_delay_log_count_ < 10) { |
| LOG(WARNING) << "Large audio delay, capture delay: " |
| << capture_delay.InMillisecondsF() |
| << "ms; playout delay: " << playout_delay.InMillisecondsF() |
| << "ms"; |
| ++large_delay_log_count_; |
| } |
| |
| audio_delay_stats_reporter_.ReportDelay(capture_delay, playout_delay); |
| |
| webrtc::AudioProcessing* ap = webrtc_audio_processing_.get(); |
| DCHECK_LE(total_delay_ms, std::numeric_limits<int>::max()); |
| ap->set_stream_delay_ms(base::saturated_cast<int>(total_delay_ms)); |
| |
| // Keep track of the maximum number of preferred channels. The number of |
| // output channels of APM can increase if preferred by the sinks, but |
| // never decrease. |
| max_num_preferred_output_channels_ = |
| std::max(max_num_preferred_output_channels_, num_preferred_channels); |
| |
| // Upscale the volume to the range expected by the WebRTC automatic gain |
| // controller. |
| #if BUILDFLAG(IS_WIN) || BUILDFLAG(IS_MAC) |
| DCHECK_LE(volume, 1.0); |
| #elif BUILDFLAG(IS_LINUX) || BUILDFLAG(IS_CHROMEOS_LACROS) || \ |
| BUILDFLAG(IS_OPENBSD) |
| // We have a special situation on Linux where the microphone volume can be |
| // "higher than maximum". The input volume slider in the sound preference |
| // allows the user to set a scaling that is higher than 100%. It means that |
| // even if the reported maximum levels is N, the actual microphone level can |
| // go up to 1.5x*N and that corresponds to a normalized |volume| of 1.5x. |
| DCHECK_LE(volume, 1.6); |
| #endif |
| // Map incoming volume range of [0.0, 1.0] to [0, 255] used by AGC. |
| // The volume can be higher than 255 on Linux, and it will be cropped to |
| // 255 since AGC does not allow values out of range. |
| const int max_analog_gain_level = media::MaxWebRtcAnalogGainLevel(); |
| int current_analog_gain_level = |
| static_cast<int>((volume * max_analog_gain_level) + 0.5); |
| current_analog_gain_level = |
| std::min(current_analog_gain_level, max_analog_gain_level); |
| DCHECK_LE(current_analog_gain_level, max_analog_gain_level); |
| |
| ap->set_stream_analog_level(current_analog_gain_level); |
| ap->set_stream_key_pressed(key_pressed); |
| |
| // Depending on how many channels the sinks prefer, the number of APM output |
| // channels is allowed to vary between 1 and the number of channels of the |
| // output format. The output format in turn depends on the input format. |
| // Example: With a stereo mic the output format will have 2 channels, and APM |
| // will produce 1 or 2 output channels depending on the sinks. |
| int num_apm_output_channels = |
| std::min(max_num_preferred_output_channels_, output_format_.channels()); |
| |
| // Limit number of apm output channels to 2 to avoid potential problems with |
| // discrete channel mapping. |
| num_apm_output_channels = std::min(num_apm_output_channels, 2); |
| |
| CHECK_GE(num_apm_output_channels, 1); |
| const webrtc::StreamConfig apm_output_config = webrtc::StreamConfig( |
| output_format_.sample_rate(), num_apm_output_channels); |
| |
| int err = ap->ProcessStream(process_ptrs, CreateStreamConfig(input_format_), |
| apm_output_config, output_ptrs); |
| DCHECK_EQ(err, 0) << "ProcessStream() error: " << err; |
| |
| // Upmix if the number of channels processed by APM is less than the number |
| // specified in the output format. Channels above stereo will be set to zero. |
| if (num_apm_output_channels < output_format_.channels()) { |
| if (num_apm_output_channels == 1) { |
| // The right channel is a copy of the left channel. Remaining channels |
| // have already been set to zero at initialization. |
| memcpy(&output_ptrs[1][0], &output_ptrs[0][0], |
| output_format_.frames_per_buffer() * sizeof(output_ptrs[0][0])); |
| } |
| } |
| |
| // Return a new mic volume, if the volume has been changed. |
| const int recommended_analog_gain_level = |
| ap->recommended_stream_analog_level(); |
| if (recommended_analog_gain_level == current_analog_gain_level) { |
| return absl::nullopt; |
| } else { |
| return static_cast<double>(recommended_analog_gain_level) / |
| media::MaxWebRtcAnalogGainLevel(); |
| } |
| } |
| |
| // Called on the owning sequence. |
| void AudioProcessor::SendLogMessage(const std::string& message) { |
| log_callback_.Run(base::StringPrintf("MSAP::%s [this=0x%" PRIXPTR "]", |
| message.c_str(), |
| reinterpret_cast<uintptr_t>(this))); |
| } |
| |
| absl::optional<AudioParameters> AudioProcessor::ComputeInputFormat( |
| const AudioParameters& device_format, |
| const AudioProcessingSettings& audio_processing_settings) { |
| const ChannelLayout channel_layout = device_format.channel_layout(); |
| |
| // The audio processor can only handle up to two channels. |
| if (channel_layout != CHANNEL_LAYOUT_MONO && |
| channel_layout != CHANNEL_LAYOUT_STEREO && |
| channel_layout != CHANNEL_LAYOUT_DISCRETE) { |
| return absl::nullopt; |
| } |
| |
| AudioParameters params( |
| device_format.format(), device_format.channel_layout_config(), |
| device_format.sample_rate(), |
| GetCaptureBufferSize( |
| audio_processing_settings.NeedWebrtcAudioProcessing(), |
| device_format)); |
| params.set_effects(device_format.effects()); |
| if (channel_layout == CHANNEL_LAYOUT_DISCRETE) { |
| DCHECK_LE(device_format.channels(), 2); |
| } |
| DVLOG(1) << params.AsHumanReadableString(); |
| CHECK(params.IsValid()); |
| return params; |
| } |
| |
| // If WebRTC audio processing is used, the default output format is fixed to the |
| // native WebRTC processing format in order to avoid rebuffering and resampling. |
| // If not, then the input format is essentially preserved. |
| // static |
| AudioParameters AudioProcessor::GetDefaultOutputFormat( |
| const AudioParameters& input_format, |
| const AudioProcessingSettings& settings) { |
| const bool need_webrtc_audio_processing = |
| settings.NeedWebrtcAudioProcessing(); |
| // TODO(crbug.com/1336055): Investigate why chromecast devices need special |
| // logic here. |
| const int output_sample_rate = |
| need_webrtc_audio_processing ? |
| #if BUILDFLAG(IS_CASTOS) || BUILDFLAG(IS_CAST_ANDROID) |
| std::min(media::kAudioProcessingSampleRateHz, |
| input_format.sample_rate()) |
| #else |
| media::kAudioProcessingSampleRateHz |
| #endif |
| : input_format.sample_rate(); |
| |
| media::ChannelLayoutConfig output_channel_layout_config; |
| if (!need_webrtc_audio_processing) { |
| output_channel_layout_config = input_format.channel_layout_config(); |
| } else if (settings.multi_channel_capture_processing) { |
| // The number of output channels is equal to the number of input channels. |
| // If the media stream audio processor receives stereo input it will |
| // output stereo. To reduce computational complexity, APM will not perform |
| // full multichannel processing unless any sink requests more than one |
| // channel. If the input is multichannel but the sinks are not interested |
| // in more than one channel, APM will internally downmix the signal to |
| // mono and process it. The processed mono signal will then be upmixed to |
| // same number of channels as the input before leaving the media stream |
| // audio processor. If a sink later requests stereo, APM will start |
| // performing true stereo processing. There will be no need to change the |
| // output format. |
| |
| output_channel_layout_config = input_format.channel_layout_config(); |
| } else { |
| output_channel_layout_config = ChannelLayoutConfig::Mono(); |
| } |
| |
| // When processing is enabled, the buffer size is dictated by |
| // webrtc::AudioProcessing (typically 10 ms). When processing is disabled, we |
| // use the same size as the source if it is less than that. |
| // |
| // TODO(ajm): This conditional buffer size appears to be assuming knowledge of |
| // the sink based on the source parameters. PeerConnection sinks seem to want |
| // 10 ms chunks regardless, while WebAudio sinks want less, and we're assuming |
| // we can identify WebAudio sinks by the input chunk size. Less fragile would |
| // be to have the sink actually tell us how much it wants (as in the above |
| // todo). |
| int output_frames = webrtc::AudioProcessing::GetFrameSize(output_sample_rate); |
| if (!need_webrtc_audio_processing && |
| input_format.frames_per_buffer() < output_frames) { |
| output_frames = input_format.frames_per_buffer(); |
| } |
| |
| media::AudioParameters output_format = media::AudioParameters( |
| input_format.format(), output_channel_layout_config, output_sample_rate, |
| output_frames); |
| return output_format; |
| } |
| |
| } // namespace media |
