Make AudioParameters a class instead of a struct

media/audio/linux/alsa_input.cc

 // Copyright (c) 2012 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 "media/audio/linux/alsa_input.h"
 
 #include "base/basictypes.h"
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/message_loop.h"
@@ skipping 11 matching lines @@
 
 const char* AlsaPcmInputStream::kAutoSelectDevice = "";
 
 AlsaPcmInputStream::AlsaPcmInputStream(AudioManagerLinux* audio_manager,
                                        const std::string& device_name,
                                        const AudioParameters& params,
                                        AlsaWrapper* wrapper)
     : audio_manager_(audio_manager),
       device_name_(device_name),
       params_(params),
-      bytes_per_packet_(params.samples_per_packet *
-                        (params.channels * params.bits_per_sample) / 8),
+      bytes_per_packet_(params.samples_per_packet() *
+                        (params.channels() * params.bits_per_sample()) / 8),
       wrapper_(wrapper),
       packet_duration_ms_(
-          (params.samples_per_packet * base::Time::kMillisecondsPerSecond) /
-          params.sample_rate),
+          (params.samples_per_packet() * base::Time::kMillisecondsPerSecond) /
+          params.samples_per_second()),
       callback_(NULL),
       device_handle_(NULL),
       mixer_handle_(NULL),
       mixer_element_handle_(NULL),
       ALLOW_THIS_IN_INITIALIZER_LIST(weak_factory_(this)),
       read_callback_behind_schedule_(false) {
 }
 
 AlsaPcmInputStream::~AlsaPcmInputStream() {}
 
 bool AlsaPcmInputStream::Open() {
   if (device_handle_)
     return false;  // Already open.
 
   snd_pcm_format_t pcm_format = alsa_util::BitsToFormat(
-      params_.bits_per_sample);
+      params_.bits_per_sample());
   if (pcm_format == SND_PCM_FORMAT_UNKNOWN) {
     LOG(WARNING) << "Unsupported bits per sample: "
-                 << params_.bits_per_sample;
+                 << params_.bits_per_sample();
     return false;
   }
 
   uint32 latency_us = packet_duration_ms_ * kNumPacketsInRingBuffer *
       base::Time::kMicrosecondsPerMillisecond;
 
   // Use the same minimum required latency as output.
   latency_us = std::max(latency_us, AlsaPcmOutputStream::kMinLatencyMicros);
 
   if (device_name_ == kAutoSelectDevice) {
     const char* device_names[] = { kDefaultDevice1, kDefaultDevice2 };
     for (size_t i = 0; i < arraysize(device_names); ++i) {
-      device_handle_ = alsa_util::OpenCaptureDevice(wrapper_, device_names[i],
-                                                    params_.channels,
-                                                    params_.sample_rate,
-                                                    pcm_format, latency_us);
+      device_handle_ = alsa_util::OpenCaptureDevice(
+          wrapper_, device_names[i], params_.channels(),
+          params_.samples_per_second(), pcm_format, latency_us);
+
       if (device_handle_) {
         device_name_ = device_names[i];
         break;
       }
     }
   } else {
     device_handle_ = alsa_util::OpenCaptureDevice(wrapper_,
                                                   device_name_.c_str(),
-                                                  params_.channels,
-                                                  params_.sample_rate,
+                                                  params_.channels(),
+                                                  params_.samples_per_second(),
                                                   pcm_format, latency_us);
   }
 
   if (device_handle_) {
     audio_packet_.reset(new uint8[bytes_per_packet_]);
 
     // Open the microphone mixer.
     mixer_handle_ = alsa_util::OpenMixer(wrapper_, device_name_);
     if (mixer_handle_) {
       mixer_element_handle_ = alsa_util::LoadCaptureMixerElement(
@@ skipping 76 matching lines @@
 
 void AlsaPcmInputStream::ReadAudio() {
   DCHECK(callback_);
 
   snd_pcm_sframes_t frames = wrapper_->PcmAvailUpdate(device_handle_);
   if (frames < 0) {  // Potentially recoverable error?
     LOG(WARNING) << "PcmAvailUpdate(): " << wrapper_->StrError(frames);
     Recover(frames);
   }
 
-  if (frames < params_.samples_per_packet) {
+  if (frames < params_.samples_per_packet()) {
     // Not enough data yet or error happened. In both cases wait for a very
     // small duration before checking again.
     // Even Though read callback was behind schedule, there is no data, so
     // reset the next_read_time_.
     if (read_callback_behind_schedule_) {
       next_read_time_ = base::Time::Now();
       read_callback_behind_schedule_ = false;
     }
 
     base::TimeDelta next_check_time = base::TimeDelta::FromMilliseconds(
         packet_duration_ms_ / 2);
     MessageLoop::current()->PostDelayedTask(
         FROM_HERE,
         base::Bind(&AlsaPcmInputStream::ReadAudio, weak_factory_.GetWeakPtr()),
         next_check_time);
     return;
   }
 
-  int num_packets = frames / params_.samples_per_packet;
+  int num_packets = frames / params_.samples_per_packet();
   int num_packets_read = num_packets;
-  int bytes_per_frame = params_.channels * params_.bits_per_sample / 8;
+  int bytes_per_frame = params_.channels() * params_.bits_per_sample() / 8;
   uint32 hardware_delay_bytes =
       static_cast<uint32>(GetCurrentDelay() * bytes_per_frame);
   while (num_packets--) {
     int frames_read = wrapper_->PcmReadi(device_handle_, audio_packet_.get(),
-                                         params_.samples_per_packet);
-    if (frames_read == params_.samples_per_packet) {
+                                         params_.samples_per_packet());
+    if (frames_read == params_.samples_per_packet()) {
       callback_->OnData(this, audio_packet_.get(), bytes_per_packet_,
                         hardware_delay_bytes);
     } else {
       LOG(WARNING) << "PcmReadi returning less than expected frames: "
-                   << frames_read << " vs. " << params_.samples_per_packet
+                   << frames_read << " vs. " << params_.samples_per_packet()
                    << ". Dropping this packet.";
     }
   }
 
   next_read_time_ += base::TimeDelta::FromMilliseconds(
       packet_duration_ms_ * num_packets_read);
   base::TimeDelta delay = next_read_time_ - base::Time::Now();
   if (delay < base::TimeDelta()) {
     LOG(WARNING) << "Audio read callback behind schedule by "
                  << (packet_duration_ms_ - delay.InMilliseconds())
@@ skipping 99 matching lines @@
     return 0.0;
   }
 
   return static_cast<double>(current_volume);
 }
 
 void AlsaPcmInputStream::HandleError(const char* method, int error) {
   LOG(WARNING) << method << ": " << wrapper_->StrError(error);
   callback_->OnError(this, error);
 }