Revert of [Chromecast] Process streams with different post-processing.

chromecast/media/cma/backend/alsa/stream_mixer_alsa.cc

Index: chromecast/media/cma/backend/alsa/stream_mixer_alsa.cc
diff --git a/chromecast/media/cma/backend/alsa/stream_mixer_alsa.cc b/chromecast/media/cma/backend/alsa/stream_mixer_alsa.cc
index e0ea0e0696b79c2b233f649b249943cc9b5eec6f..463a300aba2498a5aca63e6438cf046cc7c816c3 100644
--- a/chromecast/media/cma/backend/alsa/stream_mixer_alsa.cc
+++ b/chromecast/media/cma/backend/alsa/stream_mixer_alsa.cc
@@ -7,26 +7,20 @@
 #include <algorithm>
 #include <cmath>
 #include <limits>
-#include <unordered_set>
 #include <utility>
 
 #include "base/bind_helpers.h"
 #include "base/command_line.h"
 #include "base/lazy_instance.h"
-#include "base/memory/ptr_util.h"
 #include "base/memory/weak_ptr.h"
-#include "base/numerics/saturated_arithmetic.h"
 #include "base/single_thread_task_runner.h"
 #include "base/strings/string_number_conversions.h"
 #include "base/threading/platform_thread.h"
 #include "base/threading/thread_task_runner_handle.h"
 #include "chromecast/base/chromecast_switches.h"
-#include "chromecast/media/base/audio_device_ids.h"
 #include "chromecast/media/cma/backend/alsa/alsa_wrapper.h"
 #include "chromecast/media/cma/backend/alsa/audio_filter_factory.h"
-#include "chromecast/media/cma/backend/alsa/filter_group.h"
 #include "chromecast/media/cma/backend/alsa/stream_mixer_alsa_input_impl.h"
-#include "media/audio/audio_device_description.h"
 #include "media/base/audio_bus.h"
 #include "media/base/media_switches.h"
 
@@ -118,6 +112,9 @@
 const snd_pcm_format_t kPreferredSampleFormats[] = {SND_PCM_FORMAT_S32,
                                                     SND_PCM_FORMAT_S16};
 
+// How many seconds of silence should be passed to the filters to flush them.
+const float kSilenceSecondsToFilter = 1.0f;
+
 const int64_t kNoTimestamp = std::numeric_limits<int64_t>::min();
 
 int64_t TimespecToMicroseconds(struct timespec time) {
@@ -168,12 +165,6 @@
     return false;
   }
   return true;
-}
-
-void VectorAccumulate(const int32_t* source, size_t size, int32_t* dest) {
-  for (size_t i = 0; i < size; ++i) {
-    dest[i] = base::SaturatedAddition(source[i], dest[i]);
-  }
 }
 
 class StreamMixerAlsaInstance : public StreamMixerAlsa {
@@ -254,23 +245,11 @@
           ? kLowSampleRateCutoff
           : 0;
 
-  // Create filter groups.
-  // TODO(bshaya): Switch to filter groups based on AudioContentType
-  filter_groups_.push_back(base::MakeUnique<FilterGroup>(
-      std::unordered_set<std::string>(
-          {::media::AudioDeviceDescription::kCommunicationsDeviceId}),
-      AudioFilterFactory::COMMUNICATION_AUDIO_FILTER));
-  filter_groups_.push_back(base::MakeUnique<FilterGroup>(
-      std::unordered_set<std::string>({kAlarmAudioDeviceId}),
-      AudioFilterFactory::ALARM_AUDIO_FILTER));
-  filter_groups_.push_back(base::MakeUnique<FilterGroup>(
-      std::unordered_set<std::string>({kTtsAudioDeviceId}),
-      AudioFilterFactory::TTS_AUDIO_FILTER));
-  filter_groups_.push_back(base::MakeUnique<FilterGroup>(
-      std::unordered_set<std::string>(
-          {::media::AudioDeviceDescription::kDefaultDeviceId,
-           kEarconAudioDeviceId}),
-      AudioFilterFactory::MEDIA_AUDIO_FILTER));
+  // Create filters
+  pre_loopback_filter_ = AudioFilterFactory::MakeAudioFilter(
+      AudioFilterFactory::PRE_LOOPBACK_FILTER);
+  post_loopback_filter_ = AudioFilterFactory::MakeAudioFilter(
+      AudioFilterFactory::POST_LOOPBACK_FILTER);
 
   DefineAlsaParameters();
 }
@@ -550,9 +529,14 @@
   }
 
   // Initialize filters
-  for (auto&& filter_group : filter_groups_) {
-    filter_group->Initialize(output_samples_per_second_,
-                             ::media::SampleFormat::kSampleFormatS32);
+  if (pre_loopback_filter_) {
+    pre_loopback_filter_->SetSampleRateAndFormat(
+        output_samples_per_second_, ::media::SampleFormat::kSampleFormatS32);
+  }
+
+  if (post_loopback_filter_) {
+    post_loopback_filter_->SetSampleRateAndFormat(
+        output_samples_per_second_, ::media::SampleFormat::kSampleFormatS32);
   }
 
   RETURN_REPORT_ERROR(PcmPrepare, pcm_);
@@ -651,7 +635,6 @@
   }
 
   DCHECK(input);
-
   // If the new input is a primary one, we may need to change the output
   // sample rate to match its input sample rate.
   // We only change the output rate if it is not set to a fixed value.
@@ -661,31 +644,17 @@
   }
 
   check_close_timer_->Stop();
-  switch (state_) {
-    case kStateUninitialized:
-      requested_output_samples_per_second_ = input->input_samples_per_second();
-      Start();
-    // Fallthrough intended
-    case kStateNormalPlayback: {
-      bool found_filter_group = false;
-      input->Initialize(rendering_delay_);
-      for (auto&& filter_group : filter_groups_) {
-        if (filter_group->CanProcessInput(input.get())) {
-          found_filter_group = true;
-          input->set_filter_group(filter_group.get());
-          break;
-        }
-      }
-      DCHECK(found_filter_group) << "Could not find a filter group for "
-                                 << input->device_id();
-      inputs_.push_back(std::move(input));
-    } break;
-    case kStateError:
-      input->SignalError(StreamMixerAlsaInput::MixerError::kInternalError);
-      ignored_inputs_.push_back(std::move(input));
-      break;
-    default:
-      NOTREACHED();
+  if (state_ == kStateUninitialized) {
+    requested_output_samples_per_second_ = input->input_samples_per_second();
+    Start();
+    input->Initialize(rendering_delay_);
+    inputs_.push_back(std::move(input));
+  } else if (state_ == kStateNormalPlayback) {
+    input->Initialize(rendering_delay_);
+    inputs_.push_back(std::move(input));
+  } else {
+    input->SignalError(StreamMixerAlsaInput::MixerError::kInternalError);
+    ignored_inputs_.push_back(std::move(input));
   }
 }
 
@@ -793,8 +762,6 @@
 
 bool StreamMixerAlsa::TryWriteFrames() {
   DCHECK(mixer_task_runner_->BelongsToCurrentThread());
-  DCHECK_GE(filter_groups_.size(), 1u);
-
   if (state_ != kStateNormalPlayback) {
     return false;
   }
@@ -802,17 +769,12 @@
   const int min_frames_in_buffer =
       output_samples_per_second_ * kMinBufferedDataMs / 1000;
   int chunk_size = output_samples_per_second_ * kMaxWriteSizeMs / 1000;
-  bool is_silence = true;
-  for (auto&& filter_group : filter_groups_) {
-    filter_group->ClearActiveInputs();
-  }
+  std::vector<InputQueue*> active_inputs;
   for (auto&& input : inputs_) {
     int read_size = input->MaxReadSize();
     if (read_size > 0) {
-      DCHECK(input->filter_group());
-      input->filter_group()->AddActiveInput(input.get());
+      active_inputs.push_back(input.get());
       chunk_size = std::min(chunk_size, read_size);
-      is_silence = false;
     } else if (input->primary()) {
       if (alsa_->PcmStatus(pcm_, pcm_status_) != 0) {
         LOG(ERROR) << "Failed to get status";
@@ -839,52 +801,57 @@
     }
   }
 
-  if (is_silence) {
-    // No inputs have any data to provide. Push silence to prevent underrun.
+  if (active_inputs.empty()) {
+    // No inputs have any data to provide. Fill with silence to avoid underrun.
     chunk_size = kPreventUnderrunChunkSize;
-  }
-
-  // Mix and filter each group.
-  std::vector<uint8_t>* interleaved = nullptr;
-  for (auto&& filter_group : filter_groups_) {
-    if (filter_group->MixAndFilter(chunk_size)) {
-      if (!interleaved) {
-        interleaved = filter_group->GetInterleaved();
-      } else {
-        DCHECK_EQ(4, BytesPerOutputFormatSample());
-        VectorAccumulate(
-            reinterpret_cast<int32_t*>(filter_group->GetInterleaved()->data()),
-            chunk_size * kNumOutputChannels,
-            reinterpret_cast<int32_t*>(interleaved->data()));
-      }
-    }
-  }
-
-  if (!interleaved) {
-    // No group has any data, write empty buffer.
-    filter_groups_[0]->ClearInterleaved(chunk_size);
-    interleaved = filter_groups_[0]->GetInterleaved();
-  }
-
-  WriteMixedPcm(interleaved, chunk_size);
+    if (!mixed_ || mixed_->frames() < chunk_size) {
+      mixed_ = ::media::AudioBus::Create(kNumOutputChannels, chunk_size);
+    }
+
+    mixed_->Zero();
+    WriteMixedPcm(*mixed_, chunk_size, true /* is_silence */);
+    return true;
+  }
+
+  // If |mixed_| has not been allocated, or it is too small, allocate a buffer.
+  if (!mixed_ || mixed_->frames() < chunk_size) {
+    mixed_ = ::media::AudioBus::Create(kNumOutputChannels, chunk_size);
+  }
+
+  // If |temp_| has not been allocated, or is too small, allocate a buffer.
+  if (!temp_ || temp_->frames() < chunk_size) {
+    temp_ = ::media::AudioBus::Create(kNumOutputChannels, chunk_size);
+  }
+
+  mixed_->ZeroFramesPartial(0, chunk_size);
+
+  // Loop through active inputs, polling them for data, and mixing them.
+  for (InputQueue* input : active_inputs) {
+    input->GetResampledData(temp_.get(), chunk_size);
+    for (int c = 0; c < kNumOutputChannels; ++c) {
+      input->VolumeScaleAccumulate(c, temp_->channel(c), chunk_size,
+                                   mixed_->channel(c));
+    }
+  }
+
+  WriteMixedPcm(*mixed_, chunk_size, false /* is_silence */);
   return true;
-}
-
-size_t StreamMixerAlsa::InterleavedSize(int frames) {
-  return BytesPerOutputFormatSample() *
-         static_cast<size_t>(frames * kNumOutputChannels);
 }
 
 ssize_t StreamMixerAlsa::BytesPerOutputFormatSample() {
   return alsa_->PcmFormatSize(pcm_format_, 1);
 }
 
-void StreamMixerAlsa::WriteMixedPcm(std::vector<uint8_t>* interleaved,
-                                    int frames) {
+void StreamMixerAlsa::WriteMixedPcm(const ::media::AudioBus& mixed,
+                                    int frames, bool is_silence) {
   DCHECK(mixer_task_runner_->BelongsToCurrentThread());
   CHECK_PCM_INITIALIZED();
-  DCHECK(interleaved);
-  DCHECK_GE(interleaved->size(), InterleavedSize(frames));
+
+  size_t interleaved_size = static_cast<size_t>(frames * kNumOutputChannels) *
+                            BytesPerOutputFormatSample();
+  if (interleaved_.size() < interleaved_size) {
+    interleaved_.resize(interleaved_size);
+  }
 
   int64_t expected_playback_time;
   if (rendering_delay_.timestamp_microseconds == kNoTimestamp) {
@@ -894,10 +861,37 @@
                              rendering_delay_.delay_microseconds;
   }
 
+  mixed.ToInterleaved(frames, BytesPerOutputFormatSample(),
+                      interleaved_.data());
+
+  // Ensure that, on onset of silence, at least |kSilenceSecondsToFilter|
+  // second of audio get pushed through the filters to clear any memory.
+  bool filter_frames = true;
+  if (is_silence) {
+    int silence_frames_to_filter =
+        output_samples_per_second_ * kSilenceSecondsToFilter;
+    if (silence_frames_filtered_ < silence_frames_to_filter) {
+      silence_frames_filtered_ += frames;
+    } else {
+      filter_frames = false;
+    }
+  } else {
+    silence_frames_filtered_ = 0;
+  }
+
+  // Filter, send to observers, and post filter
+  if (pre_loopback_filter_ && filter_frames) {
+    pre_loopback_filter_->ProcessInterleaved(interleaved_.data(), frames);
+  }
+
   for (CastMediaShlib::LoopbackAudioObserver* observer : loopback_observers_) {
     observer->OnLoopbackAudio(expected_playback_time, kSampleFormatS32,
                               output_samples_per_second_, kNumOutputChannels,
-                              interleaved->data(), InterleavedSize(frames));
+                              interleaved_.data(), interleaved_size);
+  }
+
+  if (post_loopback_filter_ && filter_frames) {
+    post_loopback_filter_->ProcessInterleaved(interleaved_.data(), frames);
   }
 
   // If the PCM has been drained it will be in SND_PCM_STATE_SETUP and need
@@ -907,7 +901,7 @@
   }
 
   int frames_left = frames;
-  uint8_t* data = interleaved->data();
+  uint8_t* data = &interleaved_[0];
   while (frames_left) {
     int frames_or_error;
     while ((frames_or_error = alsa_->PcmWritei(pcm_, data, frames_left)) < 0) {