// Copyright 2015 The Cobalt Authors. All Rights Reserved.
//
// Licensed under the Apache License, Version 2.0 (the "License");
// you may not use this file except in compliance with the License.
// You may obtain a copy of the License at
//
//     http://www.apache.org/licenses/LICENSE-2.0
//
// Unless required by applicable law or agreed to in writing, software
// distributed under the License is distributed on an "AS IS" BASIS,
// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
// See the License for the specific language governing permissions and
// limitations under the License.

#include "cobalt/audio/audio_buffer_source_node.h"

#include <math.h>

#include <algorithm>

#include "cobalt/audio/audio_context.h"
#include "cobalt/audio/audio_helpers.h"
#include "cobalt/audio/audio_node_output.h"

namespace cobalt {
namespace audio {

#if defined(COBALT_MEDIA_SOURCE_2016)
typedef media::InterleavedSincResampler InterleavedSincResampler;
typedef media::ShellAudioBus ShellAudioBus;
#else   // defined(COBALT_MEDIA_SOURCE_2016)
typedef ::media::InterleavedSincResampler InterleavedSincResampler;
typedef ::media::ShellAudioBus ShellAudioBus;
#endif  // defined(COBALT_MEDIA_SOURCE_2016)

// numberOfInputs  : 0
// numberOfOutputs : 1
AudioBufferSourceNode::AudioBufferSourceNode(AudioContext* context)
    : AudioNode(context), state_(kNone), read_index_(0) {
  AudioLock::AutoLock lock(audio_lock());

  AddOutput(new AudioNodeOutput(this));
}

AudioBufferSourceNode::~AudioBufferSourceNode() {
  AudioLock::AutoLock lock(audio_lock());

  DCHECK_EQ(number_of_inputs(), 0u);
  RemoveAllOutputs();
}

void AudioBufferSourceNode::set_buffer(
    const scoped_refptr<AudioBuffer>& buffer) {
  AudioLock::AutoLock lock(audio_lock());

  buffer_ = buffer;

  if (buffer_->sample_rate() != context()->sample_rate()) {
    interleaved_resampler_ =
        std::unique_ptr<InterleavedSincResampler>(new InterleavedSincResampler(
            buffer_->sample_rate() / context()->sample_rate(),
            static_cast<int32>(buffer_->audio_bus()->channels())));
  }
}

void AudioBufferSourceNode::Start(double when, double offset,
                                  script::ExceptionState* exception_state) {
  Start(when, offset, 0, exception_state);
}

// TODO: Fully implement start and stop method. The starting time is
// based on the current time of AudioContext. We only support start at 0 and
// stop at 0 currently.
void AudioBufferSourceNode::Start(double when, double offset, double duration,
                                  script::ExceptionState* exception_state) {
  AudioLock::AutoLock lock(audio_lock());

  if (when != 0 || offset != 0 || duration != 0) {
    dom::DOMException::Raise(dom::DOMException::kInvalidStateErr,
                             exception_state);
    return;
  }

  if (state_ != kNone) {
    dom::DOMException::Raise(dom::DOMException::kInvalidStateErr,
                             exception_state);
    return;
  }
  state_ = kStarted;
}

void AudioBufferSourceNode::Stop(double when,
                                 script::ExceptionState* exception_state) {
  AudioLock::AutoLock lock(audio_lock());

  if (when != 0) {
    dom::DOMException::Raise(dom::DOMException::kInvalidStateErr,
                             exception_state);
    return;
  }

  if (state_ != kStarted) {
    dom::DOMException::Raise(dom::DOMException::kInvalidStateErr,
                             exception_state);
    return;
  }
  state_ = kStopped;
}

scoped_ptr<ShellAudioBus> AudioBufferSourceNode::PassAudioBusFromSource(
    int32 number_of_frames, SampleType sample_type, bool* finished) {
  DCHECK_GT(number_of_frames, 0);
  DCHECK(finished);

  // This is called by Audio thread.
  audio_lock()->AssertLocked();

  *finished = false;

  if (state_ == kNone || !buffer_) {
    return scoped_ptr<ShellAudioBus>();
  }

  if (state_ == kStopped ||
      (!interleaved_resampler_ && read_index_ == buffer_->length()) ||
      (interleaved_resampler_ && interleaved_resampler_->ReachedEOS())) {
    *finished = true;
    return scoped_ptr<ShellAudioBus>();
  }

  DCHECK_EQ(state_, kStarted);

  auto audio_bus = buffer_->audio_bus();
  DCHECK_EQ(sample_type, audio_bus->sample_type());

  int32 frames_to_end = buffer_->length() - read_index_;
  int32 channel_count = static_cast<int32>(audio_bus->channels());

  scoped_ptr<ShellAudioBus> result;

  if (!interleaved_resampler_) {
    int32 audio_bus_frames = std::min(number_of_frames, frames_to_end);
    if (sample_type == kSampleTypeInt16) {
      result.reset(new ShellAudioBus(
          channel_count, audio_bus_frames,
          reinterpret_cast<int16*>(audio_bus->interleaved_data()) +
              read_index_ * channel_count));
    } else {
      DCHECK_EQ(sample_type, kSampleTypeFloat32);

      result.reset(new ShellAudioBus(
          channel_count, audio_bus_frames,
          reinterpret_cast<float*>(audio_bus->interleaved_data()) +
              read_index_ * channel_count));
    }
    read_index_ += audio_bus_frames;
    return result.Pass();
  }

  // Resample audio if the audio buffer sample rate is not equal to the audio
  // context sample rate.

  // Queue frames.
  while (!interleaved_resampler_->HasEnoughData(number_of_frames)) {
    int32 frames_to_queue = static_cast<int32>(ceil(
        number_of_frames * buffer_->sample_rate() / context()->sample_rate()));

    frames_to_queue = std::min(frames_to_queue, frames_to_end);

    if (sample_type == kSampleTypeInt16) {
      int16* samples_in_int16 =
          reinterpret_cast<int16*>(audio_bus->interleaved_data()) +
          read_index_ * channel_count;
      scoped_array<float> samples_in_float(
          new float[frames_to_queue * channel_count]);
      for (int32 i = 0; i < frames_to_queue * channel_count; ++i) {
        samples_in_float[i] = ConvertSample<int16, float>(samples_in_int16[i]);
      }

      interleaved_resampler_->QueueBuffer(samples_in_float.Pass(),
                                          frames_to_queue);
    } else {
      DCHECK_EQ(sample_type, kSampleTypeFloat32);

      float* samples_in_float =
          reinterpret_cast<float*>(audio_bus->interleaved_data()) +
          read_index_ * channel_count;
      interleaved_resampler_->QueueBuffer(samples_in_float, frames_to_queue);
    }

    read_index_ += frames_to_queue;
    frames_to_end = buffer_->length() - read_index_;

    // Last time queueing buffer: signify end of stream.
    if (read_index_ == buffer_->length()) {
      interleaved_resampler_->QueueBuffer(NULL, 0);
    }
  }

  // Write resampled frames.
  if (sample_type == kSampleTypeInt16) {
    scoped_array<float> interleaved_output(
        new float[number_of_frames * channel_count]);
    interleaved_resampler_->Resample(interleaved_output.get(),
                                     number_of_frames);

    result.reset(new ShellAudioBus(channel_count, number_of_frames,
                                   kSampleTypeInt16, kStorageTypeInterleaved));
    for (int32 i = 0; i < channel_count * number_of_frames; ++i) {
      uint8* dest_ptr = result->interleaved_data() + sizeof(int16) * i;
      *reinterpret_cast<int16*>(dest_ptr) =
          ConvertSample<float, int16>(interleaved_output[i]);
    }
  } else {
    DCHECK_EQ(sample_type, kSampleTypeFloat32);

    result.reset(new ShellAudioBus(channel_count, number_of_frames,
                                   kSampleTypeFloat32,
                                   kStorageTypeInterleaved));
    interleaved_resampler_->Resample(
        reinterpret_cast<float*>(result->interleaved_data()), number_of_frames);
  }

  return result.Pass();
}

void AudioBufferSourceNode::TraceMembers(script::Tracer* tracer) {
  AudioNode::TraceMembers(tracer);

  tracer->Trace(buffer_);
}

}  // namespace audio
}  // namespace cobalt