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Unified Diff: media/audio/win/audio_low_latency_input_win.cc

Issue 344583002: Modifies AudioInputCallback::OnData and use media::AudioBus instead of plain byte vector (Relanding) (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: Added extra non-pure OnData API Created 6 years, 6 months ago
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Index: media/audio/win/audio_low_latency_input_win.cc
diff --git a/media/audio/win/audio_low_latency_input_win.cc b/media/audio/win/audio_low_latency_input_win.cc
index 0219ca4f900cbf77c10f0b2c2d0053f01eddeebc..c43ed22977c648af8e92245bd436a8cb5f58c0b0 100644
--- a/media/audio/win/audio_low_latency_input_win.cc
+++ b/media/audio/win/audio_low_latency_input_win.cc
@@ -9,6 +9,7 @@
#include "base/strings/utf_string_conversions.h"
#include "media/audio/win/audio_manager_win.h"
#include "media/audio/win/avrt_wrapper_win.h"
+#include "media/base/audio_bus.h"
using base::win::ScopedComPtr;
using base::win::ScopedCOMInitializer;
@@ -33,10 +34,9 @@ bool IsDefaultCommunicationDevice(IMMDeviceEnumerator* enumerator,
} // namespace
-WASAPIAudioInputStream::WASAPIAudioInputStream(
- AudioManagerWin* manager,
- const AudioParameters& params,
- const std::string& device_id)
+WASAPIAudioInputStream::WASAPIAudioInputStream(AudioManagerWin* manager,
+ const AudioParameters& params,
+ const std::string& device_id)
: manager_(manager),
capture_thread_(NULL),
opened_(false),
@@ -49,7 +49,8 @@ WASAPIAudioInputStream::WASAPIAudioInputStream(
device_id_(device_id),
perf_count_to_100ns_units_(0.0),
ms_to_frame_count_(0.0),
- sink_(NULL) {
+ sink_(NULL),
+ audio_bus_(media::AudioBus::Create(params)) {
DCHECK(manager_);
// Load the Avrt DLL if not already loaded. Required to support MMCSS.
@@ -436,16 +437,15 @@ void WASAPIAudioInputStream::Run() {
// size which was specified at construction.
uint32 delay_frames = static_cast<uint32>(audio_delay_frames + 0.5);
while (buffer_frame_index >= packet_size_frames_) {
- uint8* audio_data =
- reinterpret_cast<uint8*>(capture_buffer.get());
+ // Copy data to audio bus to match the OnData interface.
+ uint8* audio_data = reinterpret_cast<uint8*>(capture_buffer.get());
+ audio_bus_->FromInterleaved(
+ audio_data, audio_bus_->frames(), format_.wBitsPerSample / 8);
// Deliver data packet, delay estimation and volume level to
// the user.
- sink_->OnData(this,
- audio_data,
- packet_size_bytes_,
- delay_frames * frame_size_,
- volume);
+ sink_->OnData(
+ this, audio_bus_.get(), delay_frames * frame_size_, volume);
// Store parts of the recorded data which can't be delivered
// using the current packet size. The stored section will be used
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