Support configuring the audio buffer duration in the Pepper MediaStream API.

content/renderer/pepper/pepper_media_stream_audio_track_host.cc

 // Copyright 2014 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 "content/renderer/pepper/pepper_media_stream_audio_track_host.h"
 
 #include <algorithm>
 
 #include "base/bind.h"
 #include "base/location.h"
 #include "base/logging.h"
 #include "base/macros.h"
 #include "base/message_loop/message_loop_proxy.h"
 #include "base/numerics/safe_math.h"
 #include "ppapi/c/pp_errors.h"
 #include "ppapi/c/ppb_audio_buffer.h"
 #include "ppapi/host/dispatch_host_message.h"
 #include "ppapi/host/host_message_context.h"
 #include "ppapi/proxy/ppapi_messages.h"
 #include "ppapi/shared_impl/media_stream_audio_track_shared.h"
 #include "ppapi/shared_impl/media_stream_buffer.h"
 
 using media::AudioParameters;
 using ppapi::host::HostMessageContext;
 using ppapi::MediaStreamAudioTrackShared;
 
 namespace {
 
-// Max audio buffer duration in milliseconds.
-const uint32_t kMaxDuration = 10;
+// Audio buffer durations in milliseconds.
+const uint32_t kMinDuration = 10;
+const uint32_t kDefaultDuration = 10;
 
 const int32_t kDefaultNumberOfBuffers = 4;
 const int32_t kMaxNumberOfBuffers = 1000;  // 10 sec
 
 // Returns true if the |sample_rate| is supported in
 // |PP_AudioBuffer_SampleRate|, otherwise false.
 PP_AudioBuffer_SampleRate GetPPSampleRate(int sample_rate) {
   switch (sample_rate) {
     case 8000:
       return PP_AUDIOBUFFER_SAMPLERATE_8000;
@@ skipping 17 matching lines @@
 }
 
 }  // namespace
 
 namespace content {
 
 PepperMediaStreamAudioTrackHost::AudioSink::AudioSink(
     PepperMediaStreamAudioTrackHost* host)
     : host_(host),
       buffer_data_size_(0),
+      active_buffer_index_(-1),
+      active_buffers_generation_(0),
+      active_buffer_offset_(0),
       buffers_generation_(0),
       main_message_loop_proxy_(base::MessageLoopProxy::current()),
       weak_factory_(this),
       number_of_buffers_(kDefaultNumberOfBuffers),
-      bytes_per_second_(0) {}
+      bytes_per_second_(0),
+      user_buffer_duration_(kDefaultDuration) {}
 
 PepperMediaStreamAudioTrackHost::AudioSink::~AudioSink() {
   DCHECK_EQ(main_message_loop_proxy_, base::MessageLoopProxy::current());
 }
 
 void PepperMediaStreamAudioTrackHost::AudioSink::EnqueueBuffer(int32_t index) {
   DCHECK_EQ(main_message_loop_proxy_, base::MessageLoopProxy::current());
   DCHECK_GE(index, 0);
   DCHECK_LT(index, host_->buffer_manager()->number_of_buffers());
   base::AutoLock lock(lock_);
   buffers_.push_back(index);
 }
 
 void PepperMediaStreamAudioTrackHost::AudioSink::Configure(
-    int32_t number_of_buffers) {
+    int32_t number_of_buffers, int32_t duration) {
   DCHECK_EQ(main_message_loop_proxy_, base::MessageLoopProxy::current());
   bool changed = false;
   if (number_of_buffers != number_of_buffers_)
     changed = true;
+  if (duration != 0 && duration != user_buffer_duration_) {
+    user_buffer_duration_ = duration;
+    changed = true;
+  }
   number_of_buffers_ = number_of_buffers;
 
   // Initialize later in OnSetFormat if bytes_per_second_ is not know yet.
-  if (changed && bytes_per_second_ > 0)
+  if (changed && bytes_per_second_ > 0 && bytes_per_frame_ > 0)
     InitBuffers();
 }
 
 void PepperMediaStreamAudioTrackHost::AudioSink::SetFormatOnMainThread(
-    int bytes_per_second) {
+    int bytes_per_second, int bytes_per_frame) {
   bytes_per_second_ = bytes_per_second;
+  bytes_per_frame_ = bytes_per_frame;
   InitBuffers();
 }
 
 void PepperMediaStreamAudioTrackHost::AudioSink::InitBuffers() {
   DCHECK_EQ(main_message_loop_proxy_, base::MessageLoopProxy::current());
   {
     base::AutoLock lock(lock_);
     // Clear |buffers_|, so the audio thread will drop all incoming audio data.
     buffers_.clear();
     buffers_generation_++;
   }
+  int32_t frame_rate = bytes_per_second_ / bytes_per_frame_;
+  base::CheckedNumeric<int32_t> frames_per_buffer = user_buffer_duration_;
+  frames_per_buffer *= frame_rate;
+  frames_per_buffer /= base::Time::kMillisecondsPerSecond;
+  base::CheckedNumeric<int32_t> buffer_audio_size =
+      frames_per_buffer * bytes_per_frame_;
   // The size is slightly bigger than necessary, because 8 extra bytes are
-  // added into the struct. Also see |MediaStreamBuffer|.
-  base::CheckedNumeric<int32_t> buffer_size = bytes_per_second_;
-  buffer_size *= kMaxDuration;
-  buffer_size /= base::Time::kMillisecondsPerSecond;
+  // added into the struct. Also see |MediaStreamBuffer|. Also, the size of the
+  // buffer may be larger than requested, since the size of each buffer will be
+  // 4-byte aligned.
+  base::CheckedNumeric<int32_t> buffer_size = buffer_audio_size;
   buffer_size += sizeof(ppapi::MediaStreamBuffer::Audio);
+  DCHECK_GT(buffer_size.ValueOrDie(), 0);
 
   // We don't need to hold |lock_| during |host->InitBuffers()| call, because
   // we just cleared |buffers_| , so the audio thread will drop all incoming
   // audio data, and not use buffers in |host_|.
   bool result = host_->InitBuffers(number_of_buffers_,
                                    buffer_size.ValueOrDie(),
                                    kRead);
   // TODO(penghuang): Send PP_ERROR_NOMEMORY to plugin.
   CHECK(result);
 
   // Fill the |buffers_|, so the audio thread can continue receiving audio data.
   base::AutoLock lock(lock_);
+  output_buffer_size_ = buffer_audio_size.ValueOrDie();
   for (int32_t i = 0; i < number_of_buffers_; ++i) {
     int32_t index = host_->buffer_manager()->DequeueBuffer();
     DCHECK_GE(index, 0);
     buffers_.push_back(index);
   }
 }
 
 void PepperMediaStreamAudioTrackHost::AudioSink::
     SendEnqueueBufferMessageOnMainThread(int32_t index,
                                          int32_t buffers_generation) {
   DCHECK_EQ(main_message_loop_proxy_, base::MessageLoopProxy::current());
   // If |InitBuffers()| is called after this task being posted from the audio
   // thread, the buffer should become invalid already. We should ignore it.
   // And because only the main thread modifies the |buffers_generation_|,
   // so we don't need to lock |lock_| here (main thread).
   if (buffers_generation == buffers_generation_)
     host_->SendEnqueueBufferMessageToPlugin(index);
 }
 
 void PepperMediaStreamAudioTrackHost::AudioSink::OnData(const int16* audio_data,
                                                         int sample_rate,
                                                         int number_of_channels,
                                                         int number_of_frames) {
   DCHECK(audio_thread_checker_.CalledOnValidThread());
   DCHECK(audio_data);
   DCHECK_EQ(sample_rate, audio_params_.sample_rate());
   DCHECK_EQ(number_of_channels, audio_params_.channels());
   DCHECK_EQ(number_of_frames, audio_params_.frames_per_buffer());
 
+  const uint32_t bytes_per_frame = number_of_channels *
+      audio_params_.bits_per_sample() / 8;
+
+  int frames_remaining = number_of_frames;
+  base::TimeDelta timestamp_offset;
+
   base::AutoLock lock(lock_);
-  if (!buffers_.empty()) {
-    int index = buffers_.front();
-    buffers_.pop_front();
+  while (frames_remaining) {
+    if (active_buffers_generation_ != buffers_generation_) {
+      // Buffers have changed, so drop the active buffer.
+      active_buffer_index_ = -1;
+    }
+    if (active_buffer_index_ == -1 && !buffers_.empty()) {
+      active_buffers_generation_ = buffers_generation_;
+      active_buffer_offset_ = 0;
+      active_buffer_index_ = buffers_.front();
+      buffers_.pop_front();
+    }
+    if (active_buffer_index_ == -1) {
+      // Eek! We're dropping frames. Bad, bad, bad!
+      break;
+    }
+
     // TODO(penghuang): support re-sampling, etc.
     ppapi::MediaStreamBuffer::Audio* buffer =
-        &(host_->buffer_manager()->GetBufferPointer(index)->audio);
-    buffer->header.size = host_->buffer_manager()->buffer_size();
-    buffer->header.type = ppapi::MediaStreamBuffer::TYPE_AUDIO;
-    buffer->timestamp = timestamp_.InMillisecondsF();
-    buffer->sample_rate = static_cast<PP_AudioBuffer_SampleRate>(sample_rate);
-    buffer->number_of_channels = number_of_channels;
-    buffer->number_of_samples = number_of_channels * number_of_frames;
-    buffer->data_size = buffer_data_size_;
-    memcpy(buffer->data, audio_data, buffer_data_size_);
+        &(host_->buffer_manager()->GetBufferPointer(active_buffer_index_)
+          ->audio);
+    if (active_buffer_offset_ == 0) {
+      // The active buffer is new, so initialise the header and metadata fields.
+      buffer->header.size = host_->buffer_manager()->buffer_size();
+      buffer->header.type = ppapi::MediaStreamBuffer::TYPE_AUDIO;
+      buffer->timestamp = (timestamp_ + timestamp_offset).InMillisecondsF();
+      buffer->sample_rate = static_cast<PP_AudioBuffer_SampleRate>(sample_rate);
+      buffer->data_size = output_buffer_size_;
+      buffer->number_of_channels = number_of_channels;
+      buffer->number_of_samples = buffer->data_size / bytes_per_frame;
+    }
+    uint32_t buffer_bytes_remaining =
+        buffer->data_size - active_buffer_offset_;
+    DCHECK_EQ(buffer_bytes_remaining % bytes_per_frame, 0U);
+    uint32_t incoming_bytes_remaining = frames_remaining * bytes_per_frame;
+    uint32_t bytes_to_copy = std::min(buffer_bytes_remaining,
+                                      incoming_bytes_remaining);
+    uint32_t frames_to_copy = bytes_to_copy / bytes_per_frame;
+    DCHECK_EQ(bytes_to_copy % bytes_per_frame, 0U);
+    memcpy(buffer->data + active_buffer_offset_,
+           audio_data, bytes_to_copy);
+    active_buffer_offset_ += bytes_to_copy;
+    audio_data += bytes_to_copy / sizeof(*audio_data);
+    frames_remaining -= frames_to_copy;
+    timestamp_offset += base::TimeDelta::FromMilliseconds(
+        frames_to_copy * base::Time::kMillisecondsPerSecond / sample_rate);
 
-    main_message_loop_proxy_->PostTask(
-        FROM_HERE,
-        base::Bind(&AudioSink::SendEnqueueBufferMessageOnMainThread,
-                   weak_factory_.GetWeakPtr(),
-                   index,
-                   buffers_generation_));
+    DCHECK_LE(active_buffer_offset_, buffer->data_size);
+    if (active_buffer_offset_ == buffer->data_size) {
+      main_message_loop_proxy_->PostTask(
+          FROM_HERE,
+          base::Bind(&AudioSink::SendEnqueueBufferMessageOnMainThread,
+                     weak_factory_.GetWeakPtr(),
+                     active_buffer_index_,
+                     buffers_generation_));
+      active_buffer_index_ = -1;
+    }
   }
   timestamp_ += buffer_duration_;
 }
 
 void PepperMediaStreamAudioTrackHost::AudioSink::OnSetFormat(
     const AudioParameters& params) {
   DCHECK(params.IsValid());
-  DCHECK_LE(params.GetBufferDuration().InMilliseconds(), kMaxDuration);
+  // TODO(amistry): How do you handle the case where the user configures a
+  // duration that's shorter than the received buffer duration? One option is to
+  // double buffer, where the size of the intermediate ring buffer is at least
+  // max(user requested duration, received buffer duration). There are other
+  // ways of dealing with it, but which one is "correct"?
+  DCHECK_LE(params.GetBufferDuration().InMilliseconds(), kMinDuration);

[dmichael (off chromium), 2014/07/30 23:09:22]

Did you mean DCHECK_GE?

(Does the fact that this was passing imply the audio backend is always using
10ms currently?)

[Anand Mistry (off Chromium), 2014/07/31 00:41:21]

I meant LE. The case I'm catching is where the audio backend gives us 50ms
buffers, but we're using 4, 10ms buffers (the default). In which case, every
buffer we receive from the backend will drop at least 10ms of samples, by no
fault of the user. Arguably, in this case, there's the bigger problem of
receiving audio data slower than the user's desire, if they explicitly
configured 10ms buffers.

At the end of the day, the code will handle this case (and the DCHECK will
disappear from the release build) by dropping samples. But that sucks, and I
want to catch that (with trybot failures) before this case hits prod.
Yes. We always receive 10ms of 32KHz samples.

   DCHECK_EQ(params.bits_per_sample(), 16);
   DCHECK_NE(GetPPSampleRate(params.sample_rate()),
             PP_AUDIOBUFFER_SAMPLERATE_UNKNOWN);
 
   audio_params_ = params;
 
   // TODO(penghuang): support setting format more than once.
   buffer_duration_ = params.GetBufferDuration();
   buffer_data_size_ = params.GetBytesPerBuffer();
 
   if (original_audio_params_.IsValid()) {
     DCHECK_EQ(params.sample_rate(), original_audio_params_.sample_rate());
     DCHECK_EQ(params.bits_per_sample(),
               original_audio_params_.bits_per_sample());
     DCHECK_EQ(params.channels(), original_audio_params_.channels());
   } else {
     audio_thread_checker_.DetachFromThread();
     original_audio_params_ = params;
 
+    int bytes_per_frame = params.channels() * params.bits_per_sample() / 8;
     main_message_loop_proxy_->PostTask(
         FROM_HERE,
         base::Bind(&AudioSink::SetFormatOnMainThread,
                    weak_factory_.GetWeakPtr(),
-                   params.GetBytesPerSecond()));
+                   params.GetBytesPerSecond(),
+                   bytes_per_frame));
   }
 }
 
 PepperMediaStreamAudioTrackHost::PepperMediaStreamAudioTrackHost(
     RendererPpapiHost* host,
     PP_Instance instance,
     PP_Resource resource,
     const blink::WebMediaStreamTrack& track)
     : PepperMediaStreamTrackHostBase(host, instance, resource),
       track_(track),
@@ skipping 19 matching lines @@
 
 int32_t PepperMediaStreamAudioTrackHost::OnHostMsgConfigure(
     HostMessageContext* context,
     const MediaStreamAudioTrackShared::Attributes& attributes) {
   if (!MediaStreamAudioTrackShared::VerifyAttributes(attributes))
     return PP_ERROR_BADARGUMENT;
 
   int32_t buffers = attributes.buffers
                         ? std::min(kMaxNumberOfBuffers, attributes.buffers)
                         : kDefaultNumberOfBuffers;
-  audio_sink_.Configure(buffers);
+  audio_sink_.Configure(buffers, attributes.duration);
 
   context->reply_msg = PpapiPluginMsg_MediaStreamAudioTrack_ConfigureReply();
   return PP_OK;
 }
 
 void PepperMediaStreamAudioTrackHost::OnClose() {
   if (connected_) {
     MediaStreamAudioSink::RemoveFromAudioTrack(&audio_sink_, track_);
     connected_ = false;
   }
 }
 
 void PepperMediaStreamAudioTrackHost::OnNewBufferEnqueued() {
   int32_t index = buffer_manager()->DequeueBuffer();
   DCHECK_GE(index, 0);
   audio_sink_.EnqueueBuffer(index);
 }
 
 void PepperMediaStreamAudioTrackHost::DidConnectPendingHostToResource() {
   if (!connected_) {
     MediaStreamAudioSink::AddToAudioTrack(&audio_sink_, track_);
     connected_ = true;
   }
 }
 
 }  // namespace content