AudioConverter: Express delay in frames rather than msec.

media/base/audio_converter.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.
 //
 // AudioConverter implementation.  Uses MultiChannelSincResampler for resampling
 // audio, ChannelMixer for channel mixing, and AudioPullFifo for buffering.
 //
 // Delay estimates are provided to InputCallbacks based on the frame delay
 // information reported via the resampler and FIFO units.
 
 #include "media/base/audio_converter.h"
 
 #include <algorithm>
 
 #include "base/bind.h"
 #include "base/bind_helpers.h"
 #include "media/base/audio_bus.h"
 #include "media/base/audio_pull_fifo.h"
 #include "media/base/channel_mixer.h"
 #include "media/base/multi_channel_resampler.h"
 #include "media/base/vector_math.h"
 
 namespace media {
 
 AudioConverter::AudioConverter(const AudioParameters& input_params,
                                const AudioParameters& output_params,
                                bool disable_fifo)
     : chunk_size_(input_params.frames_per_buffer()),
       downmix_early_(false),
-      resampler_frame_delay_(0),
+      initial_frames_delayed_(0),
+      resampler_frames_delayed_(0),
+      io_sample_rate_ratio_(input_params.sample_rate() /
+                            static_cast<double>(output_params.sample_rate())),
       input_channel_count_(input_params.channels()) {
   CHECK(input_params.IsValid());
   CHECK(output_params.IsValid());
 
   // Handle different input and output channel layouts.
   if (input_params.channel_layout() != output_params.channel_layout() ||
       input_params.channels() != output_params.channels()) {
     DVLOG(1) << "Remixing channel layout from " << input_params.channel_layout()
              << " to " << output_params.channel_layout() << "; from "
              << input_params.channels() << " channels to "
              << output_params.channels() << " channels.";
     channel_mixer_.reset(new ChannelMixer(input_params, output_params));
 
     // Pare off data as early as we can for efficiency.
     downmix_early_ = input_params.channels() > output_params.channels();
   }
 
   // Only resample if necessary since it's expensive.
   if (input_params.sample_rate() != output_params.sample_rate()) {
     DVLOG(1) << "Resampling from " << input_params.sample_rate() << " to "
              << output_params.sample_rate();
     const int request_size = disable_fifo ? SincResampler::kDefaultRequestSize :
         input_params.frames_per_buffer();
-    const double io_sample_rate_ratio =
-        input_params.sample_rate() /
-        static_cast<double>(output_params.sample_rate());
     resampler_.reset(new MultiChannelResampler(
         downmix_early_ ? output_params.channels() : input_params.channels(),
-        io_sample_rate_ratio,
-        request_size,
+        io_sample_rate_ratio_, request_size,
         base::Bind(&AudioConverter::ProvideInput, base::Unretained(this))));
   }
 
-  input_frame_duration_ = base::TimeDelta::FromMicroseconds(
-      base::Time::kMicrosecondsPerSecond /
-      static_cast<double>(input_params.sample_rate()));
-  output_frame_duration_ = base::TimeDelta::FromMicroseconds(
-      base::Time::kMicrosecondsPerSecond /
-      static_cast<double>(output_params.sample_rate()));
-
   // The resampler can be configured to work with a specific request size, so a
   // FIFO is not necessary when resampling.
   if (disable_fifo || resampler_)
     return;
 
   // Since the output device may want a different buffer size than the caller
   // asked for, we need to use a FIFO to ensure that both sides read in chunk
   // sizes they're configured for.
   if (input_params.frames_per_buffer() != output_params.frames_per_buffer()) {
     DVLOG(1) << "Rebuffering from " << input_params.frames_per_buffer()
@@ skipping 35 matching lines @@
     return chunk_size_;
   return resampler_->ChunkSize();
 }
 
 void AudioConverter::PrimeWithSilence() {
   if (resampler_) {
     resampler_->PrimeWithSilence();
   }
 }
 
-void AudioConverter::ConvertWithDelay(const base::TimeDelta& initial_delay,
+void AudioConverter::ConvertWithDelay(uint32_t initial_frames_delayed,
                                       AudioBus* dest) {
-  initial_delay_ = initial_delay;
+  initial_frames_delayed_ = initial_frames_delayed;
 
   if (transform_inputs_.empty()) {
     dest->Zero();
     return;
   }
 
   // Determine if channel mixing should be done and if it should be done before
   // or after resampling.  If it's possible to reduce the channel count prior to
   // resampling we can save a lot of processing time.  Vice versa, we don't want
   // to increase the channel count prior to resampling for the same reason.
@@ skipping 18 matching lines @@
   }
 
   // Finally upmix the channels if we didn't do so earlier.
   if (needs_mixing) {
     DCHECK_EQ(temp_dest->frames(), dest->frames());
     channel_mixer_->Transform(temp_dest, dest);
   }
 }
 
 void AudioConverter::Convert(AudioBus* dest) {
-  ConvertWithDelay(base::TimeDelta::FromMilliseconds(0), dest);
+  ConvertWithDelay(0, dest);
 }
 
+// TODO - make that a uint32_t

[DaleCurtis, 2016/05/24 20:27:02]

?

[chcunningham, 2016/05/25 18:21:15]

Woops. Was a note-to-self about fifo_frame_delay, thinking I would make all the
delay counters be unsigned... probably not worth it. Its a private method and we
have lots of counters in this class that could be unsigned but currently are
just int. Deleted comment, left as int.

 void AudioConverter::SourceCallback(int fifo_frame_delay, AudioBus* dest) {
   const bool needs_downmix = channel_mixer_ && downmix_early_;
 
   if (!mixer_input_audio_bus_ ||
       mixer_input_audio_bus_->frames() != dest->frames()) {
     mixer_input_audio_bus_ =
         AudioBus::Create(input_channel_count_, dest->frames());
   }
 
   // If we're downmixing early we need a temporary AudioBus which matches
   // the the input channel count and input frame size since we're passing
   // |unmixed_audio_| directly to the |source_callback_|.
   if (needs_downmix)
     CreateUnmixedAudioIfNecessary(dest->frames());
 
   AudioBus* const temp_dest = needs_downmix ? unmixed_audio_.get() : dest;
 
   // Sanity check our inputs.
   DCHECK_EQ(temp_dest->frames(), mixer_input_audio_bus_->frames());
   DCHECK_EQ(temp_dest->channels(), mixer_input_audio_bus_->channels());
 
-  // Calculate the buffer delay for this callback.
-  base::TimeDelta buffer_delay = initial_delay_;
+  // |total_frames_delayed| is reported to the *input* source in terms of the
+  // *input* sample rate. |initial_frames_delayed_| is given in terms of the
+  // output sample rate, so we scale by sample rate ratio (in/out).
+  uint32_t total_frames_delayed =
+      std::round(initial_frames_delayed_ * io_sample_rate_ratio_);
   if (resampler_) {
-    buffer_delay += base::TimeDelta::FromMicroseconds(
-        resampler_frame_delay_ * output_frame_duration_.InMicroseconds());
+    // |resampler_frames_delayed_| tallies frames queued up inside the resampler
+    // that are already converted to the output format. Scale by ratio to get
+    // delay in terms of input sample rate.
+    total_frames_delayed +=
+        std::round(resampler_frames_delayed_ * io_sample_rate_ratio_);
+    LOG(ERROR) << __FUNCTION__

[DaleCurtis, 2016/05/24 20:27:02]

Remove?

[chcunningham, 2016/05/25 18:21:15]

Done.

+               << " reampler_frames_delayed:" << resampler_frames_delayed_
+               << " total frames_delayed:" << total_frames_delayed;
   }
   if (audio_fifo_) {
-    buffer_delay += base::TimeDelta::FromMicroseconds(
-        fifo_frame_delay * input_frame_duration_.InMicroseconds());
+    total_frames_delayed += fifo_frame_delay;
   }
 
   // If we only have a single input, avoid an extra copy.
   AudioBus* const provide_input_dest =
       transform_inputs_.size() == 1 ? temp_dest : mixer_input_audio_bus_.get();
 
   // Have each mixer render its data into an output buffer then mix the result.
   for (auto* input : transform_inputs_) {
-    const float volume = input->ProvideInput(provide_input_dest, buffer_delay);
-
+    const float volume =
+        input->ProvideInput(provide_input_dest, total_frames_delayed);
     // Optimize the most common single input, full volume case.
     if (input == transform_inputs_.front()) {
       if (volume == 1.0f) {
         if (temp_dest != provide_input_dest)
           provide_input_dest->CopyTo(temp_dest);
       } else if (volume > 0) {
         for (int i = 0; i < provide_input_dest->channels(); ++i) {
           vector_math::FMUL(
               provide_input_dest->channel(i), volume,
               provide_input_dest->frames(), temp_dest->channel(i));
@@ skipping 16 matching lines @@
     }
   }
 
   if (needs_downmix) {
     DCHECK_EQ(temp_dest->frames(), dest->frames());
     channel_mixer_->Transform(temp_dest, dest);
   }
 }
 
 void AudioConverter::ProvideInput(int resampler_frame_delay, AudioBus* dest) {
-  resampler_frame_delay_ = resampler_frame_delay;
+  resampler_frames_delayed_ = resampler_frame_delay;
   if (audio_fifo_)
     audio_fifo_->Consume(dest, dest->frames());
   else
     SourceCallback(0, dest);
 }
 
 void AudioConverter::CreateUnmixedAudioIfNecessary(int frames) {
   if (!unmixed_audio_ || unmixed_audio_->frames() != frames)
     unmixed_audio_ = AudioBus::Create(input_channel_count_, frames);
 }
 
 }  // namespace media