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authordalecurtis@chromium.org <dalecurtis@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98>2012-09-05 13:39:21 +0000
committerdalecurtis@chromium.org <dalecurtis@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98>2012-09-05 13:39:21 +0000
commitdfc48e4ce465db7e44e7cb7a4d2628958e1787a0 (patch)
treed387f8da52225eb08fe29d6bc17808d04739b26c /ppapi/shared_impl/ppb_audio_shared.cc
parent42e66e5f9dc55b57bfc95f8a49836fba59ad0cba (diff)
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Switch OnMoreData() to use AudioBus.
As titled, with this change we're now piping float data around the pipeline from end to end. This change is in preparation for browser side channel remixing and resampling. As a consequence of this change the shared memory now represents the contents of an AudioBus object, which is essentially audio data in a float planar format. BUG=114700 TEST=Should be no audible change. Ran all existing tests. Compiled ran WebAudio/HTML5/WebRTC on all platforms and PPAPI on Linux. Review URL: https://chromiumcodereview.appspot.com/10832285 git-svn-id: svn://svn.chromium.org/chrome/trunk/src@154951 0039d316-1c4b-4281-b951-d872f2087c98
Diffstat (limited to 'ppapi/shared_impl/ppb_audio_shared.cc')
-rw-r--r--ppapi/shared_impl/ppb_audio_shared.cc37
1 files changed, 31 insertions, 6 deletions
diff --git a/ppapi/shared_impl/ppb_audio_shared.cc b/ppapi/shared_impl/ppb_audio_shared.cc
index d131bb3..f275559 100644
--- a/ppapi/shared_impl/ppb_audio_shared.cc
+++ b/ppapi/shared_impl/ppb_audio_shared.cc
@@ -8,6 +8,10 @@
#include "media/audio/shared_memory_util.h"
#include "ppapi/shared_impl/ppapi_globals.h"
+// Hard coded values from PepperPlatformAudioOutputImpl.
+// TODO(dalecurtis): PPAPI shouldn't hard code these values for all clients.
+enum { kChannels = 2, kBytesPerSample = 2 };
+
namespace ppapi {
#if defined(OS_NACL)
@@ -25,7 +29,8 @@ PPB_Audio_Shared::PPB_Audio_Shared()
thread_active_(false),
#endif
callback_(NULL),
- user_data_(NULL) {
+ user_data_(NULL),
+ client_buffer_size_bytes_(0) {
}
PPB_Audio_Shared::~PPB_Audio_Shared() {
@@ -67,7 +72,8 @@ void PPB_Audio_Shared::SetStreamInfo(
PP_Instance instance,
base::SharedMemoryHandle shared_memory_handle,
size_t shared_memory_size,
- base::SyncSocket::Handle socket_handle) {
+ base::SyncSocket::Handle socket_handle,
+ int sample_frame_count) {
socket_.reset(new base::CancelableSyncSocket(socket_handle));
shared_memory_.reset(new base::SharedMemory(shared_memory_handle, false));
shared_memory_size_ = shared_memory_size;
@@ -76,6 +82,13 @@ void PPB_Audio_Shared::SetStreamInfo(
media::TotalSharedMemorySizeInBytes(shared_memory_size_))) {
PpapiGlobals::Get()->LogWithSource(instance, PP_LOGLEVEL_WARNING, "",
"Failed to map shared memory for PPB_Audio_Shared.");
+ } else {
+ audio_bus_ = media::AudioBus::WrapMemory(
+ kChannels, sample_frame_count, shared_memory_->memory());
+ // Setup integer audio buffer for user audio data.
+ client_buffer_size_bytes_ =
+ audio_bus_->frames() * audio_bus_->channels() * kBytesPerSample;
+ client_buffer_.reset(new uint8_t[client_buffer_size_bytes_]);
}
StartThread();
@@ -83,12 +96,14 @@ void PPB_Audio_Shared::SetStreamInfo(
void PPB_Audio_Shared::StartThread() {
// Don't start the thread unless all our state is set up correctly.
- if (!playing_ || !callback_ || !socket_.get() || !shared_memory_->memory())
+ if (!playing_ || !callback_ || !socket_.get() || !shared_memory_->memory() ||
+ !audio_bus_.get() || !client_buffer_.get())
return;
// Clear contents of shm buffer before starting audio thread. This will
// prevent a burst of static if for some reason the audio thread doesn't
// start up quickly enough.
memset(shared_memory_->memory(), 0, shared_memory_size_);
+ memset(client_buffer_.get(), 0, client_buffer_size_bytes_);
#if !defined(OS_NACL)
DCHECK(!audio_thread_.get());
audio_thread_.reset(new base::DelegateSimpleThread(
@@ -140,16 +155,26 @@ void PPB_Audio_Shared::CallRun(void* self) {
void PPB_Audio_Shared::Run() {
int pending_data;
- void* buffer = shared_memory_->memory();
+ const int bytes_per_frame =
+ sizeof(*audio_bus_->channel(0)) * audio_bus_->channels();
while (sizeof(pending_data) ==
socket_->Receive(&pending_data, sizeof(pending_data)) &&
pending_data != media::kPauseMark) {
- callback_(buffer, shared_memory_size_, user_data_);
+ callback_(client_buffer_.get(), client_buffer_size_bytes_, user_data_);
+
+ // Deinterleave the audio data into the shared memory as float.
+ audio_bus_->FromInterleaved(
+ client_buffer_.get(), audio_bus_->frames(), kBytesPerSample);
// Let the host know we are done.
+ // TODO(dalecurtis): Technically this is not the exact size. Due to channel
+ // padding for alignment, there may be more data available than this. We're
+ // relying on AudioSyncReader::Read() to parse this with that in mind.
+ // Rename these methods to Set/GetActualFrameCount().
media::SetActualDataSizeInBytes(
- shared_memory_.get(), shared_memory_size_, shared_memory_size_);
+ shared_memory_.get(), shared_memory_size_,
+ audio_bus_->frames() * bytes_per_frame);
}
}