Make AudioParameters a class instead of a struct

media/audio/linux/alsa_input.cc

Index: media/audio/linux/alsa_input.cc
diff --git a/media/audio/linux/alsa_input.cc b/media/audio/linux/alsa_input.cc
index 101c1985ec494cee3d29ecb75c9d8ad8f26a612a..8c53b115076de64f250af0a3cb58b72674f011e8 100644
--- a/media/audio/linux/alsa_input.cc
+++ b/media/audio/linux/alsa_input.cc
@@ -29,12 +29,12 @@ AlsaPcmInputStream::AlsaPcmInputStream(AudioManagerLinux* audio_manager,
     : 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),
@@ -50,10 +50,10 @@ bool AlsaPcmInputStream::Open() {
     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;
   }
 
@@ -66,10 +66,10 @@ bool AlsaPcmInputStream::Open() {
   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;
@@ -78,8 +78,8 @@ bool AlsaPcmInputStream::Open() {
   } 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);
   }
 
@@ -176,7 +176,7 @@ void AlsaPcmInputStream::ReadAudio() {
     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
@@ -195,20 +195,20 @@ void AlsaPcmInputStream::ReadAudio() {
     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.";
     }
   }