[Chromecast] Implement new volume control API

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

 // Copyright 2015 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "chromecast/media/cma/backend/alsa/stream_mixer_alsa.h"
 
 #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"
@@ skipping 88 matching lines @@
                                                     SND_PCM_FORMAT_S16};
 
 const int64_t kNoTimestamp = std::numeric_limits<int64_t>::min();
 
 int64_t TimespecToMicroseconds(struct timespec time) {
   return static_cast<int64_t>(time.tv_sec) *
              base::Time::kMicrosecondsPerSecond +
          time.tv_nsec / 1000;
 }
 
-bool GetSwitchValueAsInt(const std::string& switch_name,
-                         int default_value,
-                         int* value) {
-  DCHECK(value);
-  *value = default_value;
-  if (!base::CommandLine::InitializedForCurrentProcess()) {
-    LOG(WARNING) << "No CommandLine for current process.";
-    return false;
-  }
-  const base::CommandLine* command_line =
-      base::CommandLine::ForCurrentProcess();
-  if (!command_line->HasSwitch(switch_name)) {
-    return false;
-  }
-
-  int arg_value;
-  if (!base::StringToInt(command_line->GetSwitchValueASCII(switch_name),
-                         &arg_value)) {
-    LOG(DFATAL) << "--" << switch_name << " only accepts integers as arguments";
-    return false;
-  }
-  *value = arg_value;
-  return true;
-}
-
-bool GetSwitchValueAsNonNegativeInt(const std::string& switch_name,
-                                    int default_value,
-                                    int* value) {
-  DCHECK_GE(default_value, 0) << "--" << switch_name
-                              << " must have a non-negative default value";
-  DCHECK(value);
-
-  if (!GetSwitchValueAsInt(switch_name, default_value, value)) {
-    return false;
-  }
-
-  if (*value < 0) {
-    LOG(DFATAL) << "--" << switch_name << " must have a non-negative value";
-    *value = default_value;
-    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 {
  public:
   StreamMixerAlsaInstance() {}
   ~StreamMixerAlsaInstance() override {}
@@ skipping 24 matching lines @@
 StreamMixerAlsa::StreamMixerAlsa()
     : mixer_thread_(new base::Thread("ALSA CMA mixer thread")),
       mixer_task_runner_(nullptr),
       requested_output_samples_per_second_(kInvalidSampleRate),
       output_samples_per_second_(kInvalidSampleRate),
       pcm_(nullptr),
       pcm_hw_params_(nullptr),
       pcm_status_(nullptr),
       pcm_format_(SND_PCM_FORMAT_UNKNOWN),
       alsa_buffer_size_(0),
-      alsa_period_explicitly_set(false),
       alsa_period_size_(0),
       alsa_start_threshold_(0),
       alsa_avail_min_(0),
       state_(kStateUninitialized),
       retry_write_frames_timer_(new base::Timer(false, false)),
       check_close_timeout_(kDefaultCheckCloseTimeoutMs),
       check_close_timer_(new base::Timer(false, false)) {
+  for (auto type : {AudioContentType::kMedia, AudioContentType::kAlarm,

[slan, 2017/03/13 22:09:51]

Perhaps we should use a static assert in here to make sure that we don't miss
this when we add another stream type?

Or, it may just be easier to use a map and a new struct:

struct VolumeConfig {
 public:  
  VolumeConfig() : vol(0.0), limit(0.0), muted(false) {}
  float vol, limit;
  bool muted;
}

std::unordered_map<AudioConentType, VolumeConfig> volume_map_;

That way you get this initialization for free whenever you access it:
volume_map_[type].vol = vol;

[kmackay, 2017/03/14 00:20:59]

Done.

+                    AudioContentType::kCommunication}) {
+    volume_[type] = 1.0f;
+    volume_limit_[type] = 1.0f;
+    muted_[type] = false;
+  }
+
   if (single_threaded_for_test_) {
     mixer_task_runner_ = base::ThreadTaskRunnerHandle::Get();
   } else {
     base::Thread::Options options;
     options.priority = base::ThreadPriority::REALTIME_AUDIO;
     mixer_thread_->StartWithOptions(options);
     mixer_task_runner_ = mixer_thread_->task_runner();
   }
 
   alsa_device_name_ = kOutputDeviceDefaultName;
   if (base::CommandLine::InitializedForCurrentProcess() &&
       base::CommandLine::ForCurrentProcess()->HasSwitch(
           switches::kAlsaOutputDevice)) {
     alsa_device_name_ =
         base::CommandLine::ForCurrentProcess()->GetSwitchValueASCII(
             switches::kAlsaOutputDevice);
   }
 
-  int fixed_samples_per_second;
-  GetSwitchValueAsNonNegativeInt(switches::kAlsaFixedOutputSampleRate,
-                                 kInvalidSampleRate, &fixed_samples_per_second);
+  int fixed_samples_per_second = GetSwitchValueNonNegativeInt(
+      switches::kAlsaFixedOutputSampleRate, kInvalidSampleRate);
   if (fixed_samples_per_second != kInvalidSampleRate) {
     LOG(INFO) << "Setting fixed sample rate to " << fixed_samples_per_second;
   }
 
   fixed_output_samples_per_second_ = fixed_samples_per_second;
 
   low_sample_rate_cutoff_ =
       chromecast::GetSwitchValueBoolean(switches::kAlsaEnableUpsampling, false)
           ? 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));
+      AudioFilterFactory::COMMUNICATION_AUDIO_FILTER,
+      AudioContentType::kMedia));
   filter_groups_.push_back(base::MakeUnique<FilterGroup>(
       std::unordered_set<std::string>({kAlarmAudioDeviceId}),
-      AudioFilterFactory::ALARM_AUDIO_FILTER));
+      AudioFilterFactory::ALARM_AUDIO_FILTER, AudioContentType::kAlarm));
   filter_groups_.push_back(base::MakeUnique<FilterGroup>(
       std::unordered_set<std::string>({kTtsAudioDeviceId}),
-      AudioFilterFactory::TTS_AUDIO_FILTER));
+      AudioFilterFactory::TTS_AUDIO_FILTER, AudioContentType::kCommunication));
   filter_groups_.push_back(base::MakeUnique<FilterGroup>(
       std::unordered_set<std::string>(
           {::media::AudioDeviceDescription::kDefaultDeviceId,
            kLocalAudioDeviceId, ""}),
-      AudioFilterFactory::MEDIA_AUDIO_FILTER));
+      AudioFilterFactory::MEDIA_AUDIO_FILTER, AudioContentType::kMedia));
 
   DefineAlsaParameters();
 }
 
 void StreamMixerAlsa::ResetTaskRunnerForTest() {
   mixer_task_runner_ = base::ThreadTaskRunnerHandle::Get();
 }
 
 void StreamMixerAlsa::DefineAlsaParameters() {
   // Get the ALSA output configuration from the command line.
-  int buffer_size;
-  GetSwitchValueAsNonNegativeInt(switches::kAlsaOutputBufferSize,
-                                 kDefaultOutputBufferSizeFrames, &buffer_size);
-  alsa_buffer_size_ = buffer_size;
+  alsa_buffer_size_ = GetSwitchValueNonNegativeInt(
+      switches::kAlsaOutputBufferSize, kDefaultOutputBufferSizeFrames);
 
-  int period_size;
-  if (GetSwitchValueAsNonNegativeInt(switches::kAlsaOutputPeriodSize,
-                                     alsa_buffer_size_ / 16, &period_size)) {
-    if (period_size >= buffer_size) {
-      LOG(DFATAL) << "ALSA period size must be smaller than the buffer size";
-      period_size = buffer_size / 2;
-    } else {
-      alsa_period_explicitly_set = true;
-    }
+  alsa_period_size_ = GetSwitchValueNonNegativeInt(
+      switches::kAlsaOutputPeriodSize, alsa_buffer_size_ / 16);
+  if (alsa_period_size_ >= alsa_buffer_size_) {
+    LOG(DFATAL) << "ALSA period size must be smaller than the buffer size";
+    alsa_period_size_ = alsa_buffer_size_ / 2;
   }
-  alsa_period_size_ = period_size;
 
-  int start_threshold;
-  GetSwitchValueAsNonNegativeInt(switches::kAlsaOutputStartThreshold,
-                                 (buffer_size / period_size) * period_size,
-                                 &start_threshold);
-  if (start_threshold > buffer_size) {
+  alsa_start_threshold_ = GetSwitchValueNonNegativeInt(
+      switches::kAlsaOutputStartThreshold,
+      (alsa_buffer_size_ / alsa_period_size_) * alsa_period_size_);
+  if (alsa_start_threshold_ > alsa_buffer_size_) {
     LOG(DFATAL) << "ALSA start threshold must be no larger than "
                 << "the buffer size";
-    start_threshold = (buffer_size / period_size) * period_size;
+    alsa_start_threshold_ =
+        (alsa_buffer_size_ / alsa_period_size_) * alsa_period_size_;
   }
-  alsa_start_threshold_ = start_threshold;
 
   // By default, allow the transfer when at least period_size samples can be
   // processed.
-  int avail_min;
-  GetSwitchValueAsNonNegativeInt(switches::kAlsaOutputAvailMin, period_size,
-                                 &avail_min);
-  if (avail_min > buffer_size) {
+  alsa_avail_min_ = GetSwitchValueNonNegativeInt(switches::kAlsaOutputAvailMin,
+                                                 alsa_period_size_);
+  if (alsa_avail_min_ > alsa_buffer_size_) {
     LOG(DFATAL) << "ALSA avail min must be no larger than the buffer size";
-    avail_min = alsa_period_size_;
+    alsa_avail_min_ = alsa_period_size_;
   }
-  alsa_avail_min_ = avail_min;
 
   // --accept-resource-provider should imply a check close timeout of 0.
   int default_close_timeout = chromecast::GetSwitchValueBoolean(
                                   switches::kAcceptResourceProvider, false)
                                   ? 0
                                   : kDefaultCheckCloseTimeoutMs;
-  GetSwitchValueAsInt(switches::kAlsaCheckCloseTimeout, default_close_timeout,
-                      &check_close_timeout_);
+  check_close_timeout_ = GetSwitchValueInt(switches::kAlsaCheckCloseTimeout,
+                                           default_close_timeout);
 }
 
 unsigned int StreamMixerAlsa::DetermineOutputRate(unsigned int requested_rate) {
   if (fixed_output_samples_per_second_ != kInvalidSampleRate) {
     LOG(INFO) << "Requested output rate is " << requested_rate;
     LOG(INFO) << "Cannot change rate since it is fixed to "
               << fixed_output_samples_per_second_;
     return fixed_output_samples_per_second_;
   }
 
@@ skipping 95 matching lines @@
   snd_pcm_uframes_t requested_buffer_size = alsa_buffer_size_;
   RETURN_ERROR_CODE(PcmHwParamsSetBufferSizeNear, pcm_, pcm_hw_params_,
                     &alsa_buffer_size_);
   if (requested_buffer_size != alsa_buffer_size_) {
     LOG(WARNING) << "Requested buffer size (" << requested_buffer_size
                  << " frames) does not match the actual buffer size ("
                  << alsa_buffer_size_
                  << " frames). This may lead to an increase in "
                     "either audio latency or audio underruns.";
 
-    // Always try to use the value for period_size that was passed in on the
-    // command line, if any.
-    if (!alsa_period_explicitly_set) {
-      alsa_period_size_ = alsa_buffer_size_ / 16;
-    } else if (alsa_period_size_ >= alsa_buffer_size_) {
+    if (alsa_period_size_ >= alsa_buffer_size_) {
       snd_pcm_uframes_t new_period_size = alsa_buffer_size_ / 2;
       LOG(DFATAL) << "Configured period size (" << alsa_period_size_
                   << ") is >= actual buffer size (" << alsa_buffer_size_
                   << "); reducing to " << new_period_size;
       alsa_period_size_ = new_period_size;
     }
     // Scale the start threshold and avail_min based on the new buffer size.
     float original_buffer_size = static_cast<float>(requested_buffer_size);
     float avail_min_ratio = original_buffer_size / alsa_avail_min_;
     alsa_avail_min_ = alsa_buffer_size_ / avail_min_ratio;
@@ skipping 193 matching lines @@
   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.
   if (input->primary() &&
       fixed_output_samples_per_second_ == kInvalidSampleRate) {
     CheckChangeOutputRate(input->input_samples_per_second());
   }
 
+  auto type = input->content_type();
+  if (input->primary()) {
+    input->SetContentTypeVolume(std::min(volume_limit_[type], volume_[type]));
+  } else {
+    input->SetContentTypeVolume(volume_[type]);
+  }
+  input->SetMuted(muted_[type]);
+
   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_) {
@@ skipping 287 matching lines @@
     CastMediaShlib::LoopbackAudioObserver* observer) {
   RUN_ON_MIXER_THREAD(&StreamMixerAlsa::RemoveLoopbackAudioObserver, observer);
   DCHECK(std::find(loopback_observers_.begin(), loopback_observers_.end(),
                    observer) != loopback_observers_.end());
   loopback_observers_.erase(std::remove(loopback_observers_.begin(),
                                         loopback_observers_.end(), observer),
                             loopback_observers_.end());
   observer->OnRemoved();
 }
 
+void StreamMixerAlsa::SetVolume(AudioContentType type, float level) {
+  RUN_ON_MIXER_THREAD(&StreamMixerAlsa::SetVolume, type, level);
+  volume_[type] = level;
+  float effective_volume = std::min(volume_limit_[type], level);

[slan, 2017/03/13 22:09:51]

Regarding my earlier comment about using a VolumeConfig struct, you could avoid
repeating this line everywhere by implementing it in a GetEffectiveVolume()
method (ex. line 612 above, 962 below)

[kmackay, 2017/03/14 00:20:59]

Done.

+  for (auto&& input : inputs_) {
+    if (input->content_type() == type) {
+      if (input->primary()) {
+        input->SetContentTypeVolume(effective_volume);
+      } else {
+        // Volume limits don't apply to effects streams.
+        input->SetContentTypeVolume(level);
+      }
+    }
+  }
+
+  for (auto&& filter : filter_groups_) {
+    if (filter->content_type() == type) {
+      filter->set_volume(effective_volume);
+    }
+  }
+}
+
+void StreamMixerAlsa::SetMuted(AudioContentType type, bool muted) {
+  RUN_ON_MIXER_THREAD(&StreamMixerAlsa::SetMuted, type, muted);
+  muted_[type] = muted;
+  for (auto&& input : inputs_) {
+    if (input->content_type() == type) {
+      input->SetMuted(muted);
+    }
+  }
+}
+
+void StreamMixerAlsa::SetOutputLimit(AudioContentType type, float limit) {
+  RUN_ON_MIXER_THREAD(&StreamMixerAlsa::SetOutputLimit, type, limit);
+  LOG(INFO) << "Set volume limit for " << static_cast<int>(type) << " to "
+            << limit;
+  volume_limit_[type] = limit;
+  float effective_volume = std::min(volume_[type], limit);
+  for (auto&& input : inputs_) {
+    // Volume limits don't apply to effects streams.
+    if (input->primary() && input->content_type() == type) {
+      input->SetContentTypeVolume(effective_volume);
+    }
+  }
+
+  for (auto&& filter : filter_groups_) {
+    if (filter->content_type() == type) {
+      filter->set_volume(effective_volume);
+    }
+  }
+}
+
 }  // namespace media
 }  // namespace chromecast