summaryrefslogtreecommitdiffstats
path: root/chromium/media/base/audio_buffer_converter.cc
diff options
context:
space:
mode:
Diffstat (limited to 'chromium/media/base/audio_buffer_converter.cc')
-rw-r--r--chromium/media/base/audio_buffer_converter.cc249
1 files changed, 249 insertions, 0 deletions
diff --git a/chromium/media/base/audio_buffer_converter.cc b/chromium/media/base/audio_buffer_converter.cc
new file mode 100644
index 00000000000..59c66811d13
--- /dev/null
+++ b/chromium/media/base/audio_buffer_converter.cc
@@ -0,0 +1,249 @@
+// 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 "media/base/audio_buffer_converter.h"
+
+#include <cmath>
+
+#include "base/logging.h"
+#include "media/base/audio_buffer.h"
+#include "media/base/audio_bus.h"
+#include "media/base/audio_decoder_config.h"
+#include "media/base/audio_timestamp_helper.h"
+#include "media/base/buffers.h"
+#include "media/base/sinc_resampler.h"
+#include "media/base/vector_math.h"
+
+namespace media {
+
+// Is the config presented by |buffer| a config change from |params|?
+static bool IsConfigChange(const AudioParameters& params,
+ const scoped_refptr<AudioBuffer>& buffer) {
+ return buffer->sample_rate() != params.sample_rate() ||
+ buffer->channel_count() != params.channels() ||
+ buffer->channel_layout() != params.channel_layout();
+}
+
+AudioBufferConverter::AudioBufferConverter(const AudioParameters& output_params)
+ : output_params_(output_params),
+ input_params_(output_params),
+ last_input_buffer_offset_(0),
+ input_frames_(0),
+ buffered_input_frames_(0.0),
+ io_sample_rate_ratio_(1.0),
+ timestamp_helper_(output_params_.sample_rate()),
+ is_flushing_(false) {}
+
+AudioBufferConverter::~AudioBufferConverter() {}
+
+void AudioBufferConverter::AddInput(const scoped_refptr<AudioBuffer>& buffer) {
+ // On EOS flush any remaining buffered data.
+ if (buffer->end_of_stream()) {
+ Flush();
+ queued_outputs_.push_back(buffer);
+ return;
+ }
+
+ // We'll need a new |audio_converter_| if there was a config change.
+ if (IsConfigChange(input_params_, buffer))
+ ResetConverter(buffer);
+
+ // Pass straight through if there's no work to be done.
+ if (!audio_converter_) {
+ queued_outputs_.push_back(buffer);
+ return;
+ }
+
+ if (timestamp_helper_.base_timestamp() == kNoTimestamp())
+ timestamp_helper_.SetBaseTimestamp(buffer->timestamp());
+
+ queued_inputs_.push_back(buffer);
+ input_frames_ += buffer->frame_count();
+
+ ConvertIfPossible();
+}
+
+bool AudioBufferConverter::HasNextBuffer() { return !queued_outputs_.empty(); }
+
+scoped_refptr<AudioBuffer> AudioBufferConverter::GetNextBuffer() {
+ DCHECK(!queued_outputs_.empty());
+ scoped_refptr<AudioBuffer> out = queued_outputs_.front();
+ queued_outputs_.pop_front();
+ return out;
+}
+
+void AudioBufferConverter::Reset() {
+ audio_converter_.reset();
+ queued_inputs_.clear();
+ queued_outputs_.clear();
+ timestamp_helper_.SetBaseTimestamp(kNoTimestamp());
+ input_params_ = output_params_;
+ input_frames_ = 0;
+ buffered_input_frames_ = 0.0;
+ last_input_buffer_offset_ = 0;
+}
+
+void AudioBufferConverter::ResetTimestampState() {
+ Flush();
+ timestamp_helper_.SetBaseTimestamp(kNoTimestamp());
+}
+
+double AudioBufferConverter::ProvideInput(AudioBus* audio_bus,
+ base::TimeDelta buffer_delay) {
+ DCHECK(is_flushing_ || input_frames_ >= audio_bus->frames());
+
+ int requested_frames_left = audio_bus->frames();
+ int dest_index = 0;
+
+ while (requested_frames_left > 0 && !queued_inputs_.empty()) {
+ scoped_refptr<AudioBuffer> input_buffer = queued_inputs_.front();
+
+ int frames_to_read =
+ std::min(requested_frames_left,
+ input_buffer->frame_count() - last_input_buffer_offset_);
+ input_buffer->ReadFrames(
+ frames_to_read, last_input_buffer_offset_, dest_index, audio_bus);
+ last_input_buffer_offset_ += frames_to_read;
+
+ if (last_input_buffer_offset_ == input_buffer->frame_count()) {
+ // We've consumed all the frames in |input_buffer|.
+ queued_inputs_.pop_front();
+ last_input_buffer_offset_ = 0;
+ }
+
+ requested_frames_left -= frames_to_read;
+ dest_index += frames_to_read;
+ }
+
+ // If we're flushing, zero any extra space, otherwise we should always have
+ // enough data to completely fulfill the request.
+ if (is_flushing_ && requested_frames_left > 0) {
+ audio_bus->ZeroFramesPartial(audio_bus->frames() - requested_frames_left,
+ requested_frames_left);
+ } else {
+ DCHECK_EQ(requested_frames_left, 0);
+ }
+
+ input_frames_ -= audio_bus->frames() - requested_frames_left;
+ DCHECK_GE(input_frames_, 0);
+
+ buffered_input_frames_ += audio_bus->frames() - requested_frames_left;
+
+ // Full volume.
+ return 1.0;
+}
+
+void AudioBufferConverter::ResetConverter(
+ const scoped_refptr<AudioBuffer>& buffer) {
+ Flush();
+ audio_converter_.reset();
+ input_params_.Reset(
+ input_params_.format(),
+ buffer->channel_layout(),
+ buffer->channel_count(),
+ 0,
+ buffer->sample_rate(),
+ input_params_.bits_per_sample(),
+ // If resampling is needed and the FIFO disabled, the AudioConverter will
+ // always request SincResampler::kDefaultRequestSize frames. Otherwise it
+ // will use the output frame size.
+ buffer->sample_rate() == output_params_.sample_rate()
+ ? output_params_.frames_per_buffer()
+ : SincResampler::kDefaultRequestSize);
+
+ io_sample_rate_ratio_ = static_cast<double>(input_params_.sample_rate()) /
+ output_params_.sample_rate();
+
+ // If |buffer| matches |output_params_| we don't need an AudioConverter at
+ // all, and can early-out here.
+ if (!IsConfigChange(output_params_, buffer))
+ return;
+
+ // Note: The FIFO is disabled to avoid extraneous memcpy().
+ audio_converter_.reset(
+ new AudioConverter(input_params_, output_params_, true));
+ audio_converter_->AddInput(this);
+}
+
+void AudioBufferConverter::ConvertIfPossible() {
+ DCHECK(audio_converter_);
+
+ int request_frames = 0;
+
+ if (is_flushing_) {
+ // If we're flushing we want to convert *everything* even if this means
+ // we'll have to pad some silence in ProvideInput().
+ request_frames =
+ ceil((buffered_input_frames_ + input_frames_) / io_sample_rate_ratio_);
+ } else {
+ // How many calls to ProvideInput() we can satisfy completely.
+ int chunks = input_frames_ / input_params_.frames_per_buffer();
+
+ // How many output frames that corresponds to:
+ request_frames = chunks * audio_converter_->ChunkSize();
+ }
+
+ if (!request_frames)
+ return;
+
+ scoped_refptr<AudioBuffer> output_buffer =
+ AudioBuffer::CreateBuffer(kSampleFormatPlanarF32,
+ output_params_.channel_layout(),
+ output_params_.channels(),
+ output_params_.sample_rate(),
+ request_frames);
+ scoped_ptr<AudioBus> output_bus =
+ AudioBus::CreateWrapper(output_buffer->channel_count());
+
+ int frames_remaining = request_frames;
+
+ // The AudioConverter wants requests of a fixed size, so we'll slide an
+ // AudioBus of that size across the |output_buffer|.
+ while (frames_remaining != 0) {
+ // It's important that this is a multiple of AudioBus::kChannelAlignment in
+ // all requests except for the last, otherwise downstream SIMD optimizations
+ // will crash on unaligned data.
+ const int frames_this_iteration = std::min(
+ static_cast<int>(SincResampler::kDefaultRequestSize), frames_remaining);
+ const int offset_into_buffer =
+ output_buffer->frame_count() - frames_remaining;
+
+ // Wrap the portion of the AudioBuffer in an AudioBus so the AudioConverter
+ // can fill it.
+ output_bus->set_frames(frames_this_iteration);
+ for (int ch = 0; ch < output_buffer->channel_count(); ++ch) {
+ output_bus->SetChannelData(
+ ch,
+ reinterpret_cast<float*>(output_buffer->channel_data()[ch]) +
+ offset_into_buffer);
+ }
+
+ // Do the actual conversion.
+ audio_converter_->Convert(output_bus.get());
+ frames_remaining -= frames_this_iteration;
+ buffered_input_frames_ -= frames_this_iteration * io_sample_rate_ratio_;
+ }
+
+ // Compute the timestamp.
+ output_buffer->set_timestamp(timestamp_helper_.GetTimestamp());
+ timestamp_helper_.AddFrames(request_frames);
+
+ queued_outputs_.push_back(output_buffer);
+}
+
+void AudioBufferConverter::Flush() {
+ if (!audio_converter_)
+ return;
+ is_flushing_ = true;
+ ConvertIfPossible();
+ is_flushing_ = false;
+ audio_converter_->Reset();
+ DCHECK_EQ(input_frames_, 0);
+ DCHECK_EQ(last_input_buffer_offset_, 0);
+ DCHECK_LT(buffered_input_frames_, 1.0);
+ DCHECK(queued_inputs_.empty());
+ buffered_input_frames_ = 0.0;
+}
+
+} // namespace media