Supervised user whitelists: Cleanup

[chromium-blink-merge.git] / media / cast / sender / audio_encoder.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 "media/cast/sender/audio_encoder.h"

#include <algorithm>
#include <limits>
#include <string>

#include "base/bind.h"
#include "base/bind_helpers.h"
#include "base/location.h"
#include "base/stl_util.h"
#include "base/sys_byteorder.h"
#include "base/time/time.h"
#include "media/base/audio_bus.h"
#include "media/cast/cast_defines.h"
#include "media/cast/cast_environment.h"

#if !defined(OS_IOS)
#include "third_party/opus/src/include/opus.h"
#endif

#if defined(OS_MACOSX)
#include <AudioToolbox/AudioToolbox.h>
#endif

namespace media {
namespace cast {

namespace {

const int kUnderrunSkipThreshold = 3;
const int kDefaultFramesPerSecond = 100;

}  // namespace

// Base class that handles the common problem of feeding one or more AudioBus'
// data into a buffer and then, once the buffer is full, encoding the signal and
// emitting an EncodedFrame via the FrameEncodedCallback.
//
// Subclasses complete the implementation by handling the actual encoding
// details.
class AudioEncoder::ImplBase
    : public base::RefCountedThreadSafe<AudioEncoder::ImplBase> {
 public:
  ImplBase(const scoped_refptr<CastEnvironment>& cast_environment,
           Codec codec,
           int num_channels,
           int sampling_rate,
           int samples_per_frame,
           const FrameEncodedCallback& callback)
      : cast_environment_(cast_environment),
        codec_(codec),
        num_channels_(num_channels),
        samples_per_frame_(samples_per_frame),
        callback_(callback),
        operational_status_(STATUS_UNINITIALIZED),
        frame_duration_(base::TimeDelta::FromMicroseconds(
            base::Time::kMicrosecondsPerSecond * samples_per_frame_ /
            sampling_rate)),
        buffer_fill_end_(0),
        frame_id_(0),
        frame_rtp_timestamp_(0),
        samples_dropped_from_buffer_(0) {
    // Support for max sampling rate of 48KHz, 2 channels, 100 ms duration.
    const int kMaxSamplesTimesChannelsPerFrame = 48 * 2 * 100;
    if (num_channels_ <= 0 || samples_per_frame_ <= 0 ||
        frame_duration_ == base::TimeDelta() ||
        samples_per_frame_ * num_channels_ > kMaxSamplesTimesChannelsPerFrame) {
      operational_status_ = STATUS_INVALID_CONFIGURATION;
    }
  }

  OperationalStatus InitializationResult() const {
    return operational_status_;
  }

  int samples_per_frame() const {
    return samples_per_frame_;
  }

  base::TimeDelta frame_duration() const { return frame_duration_; }

  void EncodeAudio(scoped_ptr<AudioBus> audio_bus,
                   const base::TimeTicks& recorded_time) {
    DCHECK_EQ(operational_status_, STATUS_INITIALIZED);
    DCHECK(!recorded_time.is_null());

    // Determine whether |recorded_time| is consistent with the amount of audio
    // data having been processed in the past.  Resolve the underrun problem by
    // dropping data from the internal buffer and skipping ahead the next
    // frame's RTP timestamp by the estimated number of frames missed.  On the
    // other hand, don't attempt to resolve overruns: A receiver should
    // gracefully deal with an excess of audio data.
    base::TimeDelta buffer_fill_duration =
        buffer_fill_end_ * frame_duration_ / samples_per_frame_;
    if (!frame_capture_time_.is_null()) {
      const base::TimeDelta amount_ahead_by =
          recorded_time - (frame_capture_time_ + buffer_fill_duration);
      const int64 num_frames_missed = amount_ahead_by / frame_duration_;
      if (num_frames_missed > kUnderrunSkipThreshold) {
        samples_dropped_from_buffer_ += buffer_fill_end_;
        buffer_fill_end_ = 0;
        buffer_fill_duration = base::TimeDelta();
        frame_rtp_timestamp_ +=
            static_cast<uint32>(num_frames_missed * samples_per_frame_);
        DVLOG(1) << "Skipping RTP timestamp ahead to account for "
                 << num_frames_missed * samples_per_frame_
                 << " samples' worth of underrun.";
      }
    }
    frame_capture_time_ = recorded_time - buffer_fill_duration;

    // Encode all audio in |audio_bus| into zero or more frames.
    int src_pos = 0;
    while (src_pos < audio_bus->frames()) {
      const int num_samples_to_xfer = std::min(
          samples_per_frame_ - buffer_fill_end_, audio_bus->frames() - src_pos);
      DCHECK_EQ(audio_bus->channels(), num_channels_);
      TransferSamplesIntoBuffer(
          audio_bus.get(), src_pos, buffer_fill_end_, num_samples_to_xfer);
      src_pos += num_samples_to_xfer;
      buffer_fill_end_ += num_samples_to_xfer;

      if (buffer_fill_end_ < samples_per_frame_)
        break;

      scoped_ptr<EncodedFrame> audio_frame(
          new EncodedFrame());
      audio_frame->dependency = EncodedFrame::KEY;
      audio_frame->frame_id = frame_id_;
      audio_frame->referenced_frame_id = frame_id_;
      audio_frame->rtp_timestamp = frame_rtp_timestamp_;
      audio_frame->reference_time = frame_capture_time_;

      if (EncodeFromFilledBuffer(&audio_frame->data)) {
        cast_environment_->PostTask(
            CastEnvironment::MAIN,
            FROM_HERE,
            base::Bind(callback_,
                       base::Passed(&audio_frame),
                       samples_dropped_from_buffer_));
        samples_dropped_from_buffer_ = 0;
      }

      // Reset the internal buffer, frame ID, and timestamps for the next frame.
      buffer_fill_end_ = 0;
      ++frame_id_;
      frame_rtp_timestamp_ += samples_per_frame_;
      frame_capture_time_ += frame_duration_;
    }
  }

 protected:
  friend class base::RefCountedThreadSafe<ImplBase>;
  virtual ~ImplBase() {}

  virtual void TransferSamplesIntoBuffer(const AudioBus* audio_bus,
                                         int source_offset,
                                         int buffer_fill_offset,
                                         int num_samples) = 0;
  virtual bool EncodeFromFilledBuffer(std::string* out) = 0;

  const scoped_refptr<CastEnvironment> cast_environment_;
  const Codec codec_;
  const int num_channels_;
  const int samples_per_frame_;
  const FrameEncodedCallback callback_;

  // Subclass' ctor is expected to set this to STATUS_INITIALIZED.
  OperationalStatus operational_status_;

  // The duration of one frame of encoded audio samples. Derived from
  // |samples_per_frame_| and the sampling rate.
  const base::TimeDelta frame_duration_;

 private:
  // In the case where a call to EncodeAudio() cannot completely fill the
  // buffer, this points to the position at which to populate data in a later
  // call.
  int buffer_fill_end_;

  // A counter used to label EncodedFrames.
  uint32 frame_id_;

  // The RTP timestamp for the next frame of encoded audio.  This is defined as
  // the number of audio samples encoded so far, plus the estimated number of
  // samples that were missed due to data underruns.  A receiver uses this value
  // to detect gaps in the audio signal data being provided.  Per the spec, RTP
  // timestamp values are allowed to overflow and roll around past zero.
  uint32 frame_rtp_timestamp_;

  // The local system time associated with the start of the next frame of
  // encoded audio.  This value is passed on to a receiver as a reference clock
  // timestamp for the purposes of synchronizing audio and video.  Its
  // progression is expected to drift relative to the elapsed time implied by
  // the RTP timestamps.
  base::TimeTicks frame_capture_time_;

  // Set to non-zero to indicate the next output frame skipped over audio
  // samples in order to recover from an input underrun.
  int samples_dropped_from_buffer_;

  DISALLOW_COPY_AND_ASSIGN(ImplBase);
};

#if !defined(OS_IOS)
class AudioEncoder::OpusImpl : public AudioEncoder::ImplBase {
 public:
  OpusImpl(const scoped_refptr<CastEnvironment>& cast_environment,
           int num_channels,
           int sampling_rate,
           int bitrate,
           const FrameEncodedCallback& callback)
      : ImplBase(cast_environment,
                 CODEC_AUDIO_OPUS,
                 num_channels,
                 sampling_rate,
                 sampling_rate / kDefaultFramesPerSecond, /* 10 ms frames */
                 callback),
        encoder_memory_(new uint8[opus_encoder_get_size(num_channels)]),
        opus_encoder_(reinterpret_cast<OpusEncoder*>(encoder_memory_.get())),
        buffer_(new float[num_channels * samples_per_frame_]) {
    if (ImplBase::operational_status_ != STATUS_UNINITIALIZED ||
        sampling_rate % samples_per_frame_ != 0 ||
        !IsValidFrameDuration(frame_duration_)) {
      return;
    }
    if (opus_encoder_init(opus_encoder_,
                          sampling_rate,
                          num_channels,
                          OPUS_APPLICATION_AUDIO) != OPUS_OK) {
      ImplBase::operational_status_ = STATUS_INVALID_CONFIGURATION;
      return;
    }
    ImplBase::operational_status_ = STATUS_INITIALIZED;

    if (bitrate <= 0) {
      // Note: As of 2013-10-31, the encoder in "auto bitrate" mode would use a
      // variable bitrate up to 102kbps for 2-channel, 48 kHz audio and a 10 ms
      // frame size.  The opus library authors may, of course, adjust this in
      // later versions.
      bitrate = OPUS_AUTO;
    }
    CHECK_EQ(opus_encoder_ctl(opus_encoder_, OPUS_SET_BITRATE(bitrate)),
             OPUS_OK);
  }

 private:
  ~OpusImpl() override {}

  void TransferSamplesIntoBuffer(const AudioBus* audio_bus,
                                 int source_offset,
                                 int buffer_fill_offset,
                                 int num_samples) override {
    // Opus requires channel-interleaved samples in a single array.
    for (int ch = 0; ch < audio_bus->channels(); ++ch) {
      const float* src = audio_bus->channel(ch) + source_offset;
      const float* const src_end = src + num_samples;
      float* dest = buffer_.get() + buffer_fill_offset * num_channels_ + ch;
      for (; src < src_end; ++src, dest += num_channels_)
        *dest = *src;
    }
  }

  bool EncodeFromFilledBuffer(std::string* out) override {
    out->resize(kOpusMaxPayloadSize);
    const opus_int32 result =
        opus_encode_float(opus_encoder_,
                          buffer_.get(),
                          samples_per_frame_,
                          reinterpret_cast<uint8*>(string_as_array(out)),
                          kOpusMaxPayloadSize);
    if (result > 1) {
      out->resize(result);
      return true;
    } else if (result < 0) {
      LOG(ERROR) << "Error code from opus_encode_float(): " << result;
      return false;
    } else {
      // Do nothing: The documentation says that a return value of zero or
      // one byte means the packet does not need to be transmitted.
      return false;
    }
  }

  static bool IsValidFrameDuration(base::TimeDelta duration) {
    // See https://tools.ietf.org/html/rfc6716#section-2.1.4
    return duration == base::TimeDelta::FromMicroseconds(2500) ||
           duration == base::TimeDelta::FromMilliseconds(5) ||
           duration == base::TimeDelta::FromMilliseconds(10) ||
           duration == base::TimeDelta::FromMilliseconds(20) ||
           duration == base::TimeDelta::FromMilliseconds(40) ||
           duration == base::TimeDelta::FromMilliseconds(60);
  }

  const scoped_ptr<uint8[]> encoder_memory_;
  OpusEncoder* const opus_encoder_;
  const scoped_ptr<float[]> buffer_;

  // This is the recommended value, according to documentation in
  // third_party/opus/src/include/opus.h, so that the Opus encoder does not
  // degrade the audio due to memory constraints.
  //
  // Note: Whereas other RTP implementations do not, the cast library is
  // perfectly capable of transporting larger than MTU-sized audio frames.
  static const int kOpusMaxPayloadSize = 4000;

  DISALLOW_COPY_AND_ASSIGN(OpusImpl);
};
#endif

#if defined(OS_MACOSX)
class AudioEncoder::AppleAacImpl : public AudioEncoder::ImplBase {
  // AAC-LC has two access unit sizes (960 and 1024). The Apple encoder only
  // supports the latter.
  static const int kAccessUnitSamples = 1024;

  // Size of an ADTS header (w/o checksum). See
  // http://wiki.multimedia.cx/index.php?title=ADTS
  static const int kAdtsHeaderSize = 7;

 public:
  AppleAacImpl(const scoped_refptr<CastEnvironment>& cast_environment,
               int num_channels,
               int sampling_rate,
               int bitrate,
               const FrameEncodedCallback& callback)
      : ImplBase(cast_environment,
                 CODEC_AUDIO_AAC,
                 num_channels,
                 sampling_rate,
                 kAccessUnitSamples,
                 callback),
        input_buffer_(AudioBus::Create(num_channels, kAccessUnitSamples)),
        input_bus_(AudioBus::CreateWrapper(num_channels)),
        max_access_unit_size_(0),
        output_buffer_(nullptr),
        converter_(nullptr),
        file_(nullptr),
        num_access_units_(0),
        can_resume_(true) {
    if (ImplBase::operational_status_ != STATUS_UNINITIALIZED) {
      return;
    }
    if (!Initialize(sampling_rate, bitrate)) {
      ImplBase::operational_status_ = STATUS_INVALID_CONFIGURATION;
      return;
    }
    ImplBase::operational_status_ = STATUS_INITIALIZED;
  }

 private:
  ~AppleAacImpl() override { Teardown(); }

  // Destroys the existing audio converter and file, if any.
  void Teardown() {
    if (converter_) {
      AudioConverterDispose(converter_);
      converter_ = nullptr;
    }
    if (file_) {
      AudioFileClose(file_);
      file_ = nullptr;
    }
  }

  // Initializes the audio converter and file. Calls Teardown to destroy any
  // existing state. This is so that Initialize() may be called to setup another
  // converter after a non-resumable interruption.
  bool Initialize(int sampling_rate, int bitrate) {
    // Teardown previous audio converter and file.
    Teardown();

    // Input data comes from AudioBus objects, which carry non-interleaved
    // packed native-endian float samples. Note that in Core Audio, a frame is
    // one sample across all channels at a given point in time. When describing
    // a non-interleaved samples format, the "per frame" fields mean "per
    // channel" or "per stream", with the exception of |mChannelsPerFrame|. For
    // uncompressed formats, one packet contains one frame.
    AudioStreamBasicDescription in_asbd;
    in_asbd.mSampleRate = sampling_rate;
    in_asbd.mFormatID = kAudioFormatLinearPCM;
    in_asbd.mFormatFlags =
        kAudioFormatFlagsNativeFloatPacked | kAudioFormatFlagIsNonInterleaved;
    in_asbd.mChannelsPerFrame = num_channels_;
    in_asbd.mBitsPerChannel = sizeof(float) * 8;
    in_asbd.mFramesPerPacket = 1;
    in_asbd.mBytesPerPacket = in_asbd.mBytesPerFrame = sizeof(float);
    in_asbd.mReserved = 0;

    // Request AAC-LC encoding, with no downmixing or downsampling.
    AudioStreamBasicDescription out_asbd;
    memset(&out_asbd, 0, sizeof(AudioStreamBasicDescription));
    out_asbd.mSampleRate = sampling_rate;
    out_asbd.mFormatID = kAudioFormatMPEG4AAC;
    out_asbd.mChannelsPerFrame = num_channels_;
    UInt32 prop_size = sizeof(out_asbd);
    if (AudioFormatGetProperty(kAudioFormatProperty_FormatInfo,
                               0,
                               nullptr,
                               &prop_size,
                               &out_asbd) != noErr) {
      return false;
    }

    if (AudioConverterNew(&in_asbd, &out_asbd, &converter_) != noErr) {
      return false;
    }

    // The converter will fully specify the output format and update the
    // relevant fields of the structure, which we can now query.
    prop_size = sizeof(out_asbd);
    if (AudioConverterGetProperty(converter_,
                                  kAudioConverterCurrentOutputStreamDescription,
                                  &prop_size,
                                  &out_asbd) != noErr) {
      return false;
    }

    // If bitrate is <= 0, allow the encoder to pick a suitable value.
    // Otherwise, set the bitrate (which can fail if the value is not suitable
    // or compatible with the output sampling rate or channels).
    if (bitrate > 0) {
      prop_size = sizeof(int);
      if (AudioConverterSetProperty(
              converter_, kAudioConverterEncodeBitRate, prop_size, &bitrate) !=
          noErr) {
        return false;
      }
    }

#if defined(OS_IOS)
    // See the comment next to |can_resume_| for details on resumption. Some
    // converters can return kAudioConverterErr_PropertyNotSupported, in which
    // case resumption is implicitly supported. This is the only location where
    // the implementation modifies |can_resume_|.
    uint32_t can_resume;
    prop_size = sizeof(can_resume);
    OSStatus oserr = AudioConverterGetProperty(
        converter_,
        kAudioConverterPropertyCanResumeFromInterruption,
        &prop_size,
        &can_resume);
    if (oserr == noErr) {
      const_cast<bool&>(can_resume_) = can_resume != 0;
    }
#endif

    // Figure out the maximum size of an access unit that the encoder can
    // produce. |mBytesPerPacket| will be 0 for variable size configurations,
    // in which case we must query the value.
    uint32_t max_access_unit_size = out_asbd.mBytesPerPacket;
    if (max_access_unit_size == 0) {
      prop_size = sizeof(max_access_unit_size);
      if (AudioConverterGetProperty(
              converter_,
              kAudioConverterPropertyMaximumOutputPacketSize,
              &prop_size,
              &max_access_unit_size) != noErr) {
        return false;
      }
    }

    // This is the only location where the implementation modifies
    // |max_access_unit_size_|.
    const_cast<uint32_t&>(max_access_unit_size_) = max_access_unit_size;

    // Allocate a buffer to store one access unit. This is the only location
    // where the implementation modifies |access_unit_buffer_|.
    const_cast<scoped_ptr<uint8[]>&>(access_unit_buffer_)
        .reset(new uint8[max_access_unit_size]);

    // Initialize the converter ABL. Note that the buffer size has to be set
    // before every encode operation, since the field is modified to indicate
    // the size of the output data (on input it indicates the buffer capacity).
    converter_abl_.mNumberBuffers = 1;
    converter_abl_.mBuffers[0].mNumberChannels = num_channels_;
    converter_abl_.mBuffers[0].mData = access_unit_buffer_.get();

    // The "magic cookie" is an encoder state vector required for decoding and
    // packetization. It is queried now from |converter_| then set on |file_|
    // after initialization.
    UInt32 cookie_size;
    if (AudioConverterGetPropertyInfo(converter_,
                                      kAudioConverterCompressionMagicCookie,
                                      &cookie_size,
                                      nullptr) != noErr) {
      return false;
    }
    scoped_ptr<uint8[]> cookie_data(new uint8[cookie_size]);
    if (AudioConverterGetProperty(converter_,
                                  kAudioConverterCompressionMagicCookie,
                                  &cookie_size,
                                  cookie_data.get()) != noErr) {
      return false;
    }

    if (AudioFileInitializeWithCallbacks(this,
                                         nullptr,
                                         &FileWriteCallback,
                                         nullptr,
                                         nullptr,
                                         kAudioFileAAC_ADTSType,
                                         &out_asbd,
                                         0,
                                         &file_) != noErr) {
      return false;
    }

    if (AudioFileSetProperty(file_,
                             kAudioFilePropertyMagicCookieData,
                             cookie_size,
                             cookie_data.get()) != noErr) {
      return false;
    }

    // Initially the input bus points to the input buffer. See the comment on
    // |input_bus_| for more on this optimization.
    input_bus_->set_frames(kAccessUnitSamples);
    for (int ch = 0; ch < input_buffer_->channels(); ++ch) {
      input_bus_->SetChannelData(ch, input_buffer_->channel(ch));
    }

    return true;
  }

  void TransferSamplesIntoBuffer(const AudioBus* audio_bus,
                                 int source_offset,
                                 int buffer_fill_offset,
                                 int num_samples) override {
    DCHECK_EQ(audio_bus->channels(), input_buffer_->channels());

    // See the comment on |input_bus_| for more on this optimization. Note that
    // we cannot elide the copy if the source offset would result in an
    // unaligned pointer.
    if (num_samples == kAccessUnitSamples &&
        source_offset * sizeof(float) % AudioBus::kChannelAlignment == 0) {
      DCHECK_EQ(buffer_fill_offset, 0);
      for (int ch = 0; ch < audio_bus->channels(); ++ch) {
        auto samples = const_cast<float*>(audio_bus->channel(ch));
        input_bus_->SetChannelData(ch, samples + source_offset);
      }
      return;
    }

    // Copy the samples into the input buffer.
    DCHECK_EQ(input_bus_->channel(0), input_buffer_->channel(0));
    audio_bus->CopyPartialFramesTo(
        source_offset, num_samples, buffer_fill_offset, input_buffer_.get());
  }

  bool EncodeFromFilledBuffer(std::string* out) override {
    // Reset the buffer size field to the buffer capacity.
    converter_abl_.mBuffers[0].mDataByteSize = max_access_unit_size_;

    // Encode the current input buffer. This is a sychronous call.
    OSStatus oserr;
    UInt32 io_num_packets = 1;
    AudioStreamPacketDescription packet_description;
    oserr = AudioConverterFillComplexBuffer(converter_,
                                            &ConverterFillDataCallback,
                                            this,
                                            &io_num_packets,
                                            &converter_abl_,
                                            &packet_description);
    if (oserr != noErr || io_num_packets == 0) {
      return false;
    }

    // Reserve space in the output buffer to write the packet.
    out->reserve(packet_description.mDataByteSize + kAdtsHeaderSize);

    // Set the current output buffer and emit an ADTS-wrapped AAC access unit.
    // This is a synchronous call. After it returns, reset the output buffer.
    output_buffer_ = out;
    oserr = AudioFileWritePackets(file_,
                                  false,
                                  converter_abl_.mBuffers[0].mDataByteSize,
                                  &packet_description,
                                  num_access_units_,
                                  &io_num_packets,
                                  converter_abl_.mBuffers[0].mData);
    output_buffer_ = nullptr;
    if (oserr != noErr || io_num_packets == 0) {
      return false;
    }
    num_access_units_ += io_num_packets;
    return true;
  }

  // The |AudioConverterFillComplexBuffer| input callback function. Configures
  // the provided |AudioBufferList| to alias |input_bus_|. The implementation
  // can only supply |kAccessUnitSamples| samples as a result of not copying
  // samples or tracking read and write positions. Note that this function is
  // called synchronously by |AudioConverterFillComplexBuffer|.
  static OSStatus ConverterFillDataCallback(
      AudioConverterRef in_converter,
      UInt32* io_num_packets,
      AudioBufferList* io_data,
      AudioStreamPacketDescription** out_packet_desc,
      void* in_encoder) {
    DCHECK(in_encoder);
    auto encoder = reinterpret_cast<AppleAacImpl*>(in_encoder);
    auto input_buffer = encoder->input_buffer_.get();
    auto input_bus = encoder->input_bus_.get();

    DCHECK_EQ(static_cast<int>(*io_num_packets), kAccessUnitSamples);
    DCHECK_EQ(io_data->mNumberBuffers,
              static_cast<unsigned>(input_bus->channels()));
    for (int i_buf = 0, end = io_data->mNumberBuffers; i_buf < end; ++i_buf) {
      io_data->mBuffers[i_buf].mNumberChannels = 1;
      io_data->mBuffers[i_buf].mDataByteSize = sizeof(float) * *io_num_packets;
      io_data->mBuffers[i_buf].mData = input_bus->channel(i_buf);

      // Reset the input bus back to the input buffer. See the comment on
      // |input_bus_| for more on this optimization.
      input_bus->SetChannelData(i_buf, input_buffer->channel(i_buf));
    }
    return noErr;
  }

  // The AudioFile write callback function. Appends the data to the encoder's
  // current |output_buffer_|.
  static OSStatus FileWriteCallback(void* in_encoder,
                                    SInt64 in_position,
                                    UInt32 in_size,
                                    const void* in_buffer,
                                    UInt32* out_size) {
    DCHECK(in_encoder);
    DCHECK(in_buffer);
    auto encoder = reinterpret_cast<const AppleAacImpl*>(in_encoder);
    auto buffer = reinterpret_cast<const std::string::value_type*>(in_buffer);

    std::string* const output_buffer = encoder->output_buffer_;
    DCHECK(output_buffer);

    output_buffer->append(buffer, in_size);
    *out_size = in_size;
    return noErr;
  }

  // Buffer that holds one AAC access unit worth of samples. The input callback
  // function provides samples from this buffer via |input_bus_| to the encoder.
  const scoped_ptr<AudioBus> input_buffer_;

  // Wrapper AudioBus used by the input callback function. Normally it wraps
  // |input_buffer_|. However, as an optimization when the client submits a
  // buffer containing exactly one access unit worth of samples, the bus is
  // redirected to the client buffer temporarily. We know that the base
  // implementation will call us right after to encode the buffer and thus we
  // can eliminate the copy into |input_buffer_|.
  const scoped_ptr<AudioBus> input_bus_;

  // A buffer that holds one AAC access unit. Initialized in |Initialize| once
  // the maximum access unit size is known.
  const scoped_ptr<uint8[]> access_unit_buffer_;

  // The maximum size of an access unit that the encoder can emit.
  const uint32_t max_access_unit_size_;

  // A temporary pointer to the current output buffer. Only non-null when
  // writing an access unit. Accessed by the AudioFile write callback function.
  std::string* output_buffer_;

  // The |AudioConverter| is responsible for AAC encoding. This is a Core Audio
  // object, not to be confused with |media::AudioConverter|.
  AudioConverterRef converter_;

  // The |AudioFile| is responsible for ADTS packetization.
  AudioFileID file_;

  // An |AudioBufferList| passed to the converter to store encoded samples.
  AudioBufferList converter_abl_;

  // The number of access units emitted so far by the encoder.
  uint64_t num_access_units_;

  // On iOS, audio codecs can be interrupted by other services (such as an
  // audio alert or phone call). Depending on the underlying hardware and
  // configuration, the codec may have to be thrown away and re-initialized
  // after such an interruption. This flag tracks if we can resume or not from
  // such an interruption. It is initialized to true, which is the only possible
  // value on OS X and on most modern iOS hardware.
  // TODO(jfroy): Implement encoder re-initialization after interruption.
  //              https://crbug.com/424787
  const bool can_resume_;

  DISALLOW_COPY_AND_ASSIGN(AppleAacImpl);
};
#endif  // defined(OS_MACOSX)

class AudioEncoder::Pcm16Impl : public AudioEncoder::ImplBase {
 public:
  Pcm16Impl(const scoped_refptr<CastEnvironment>& cast_environment,
            int num_channels,
            int sampling_rate,
            const FrameEncodedCallback& callback)
      : ImplBase(cast_environment,
                 CODEC_AUDIO_PCM16,
                 num_channels,
                 sampling_rate,
                 sampling_rate / kDefaultFramesPerSecond, /* 10 ms frames */
                 callback),
        buffer_(new int16[num_channels * samples_per_frame_]) {
    if (ImplBase::operational_status_ != STATUS_UNINITIALIZED)
      return;
    operational_status_ = STATUS_INITIALIZED;
  }

 private:
  ~Pcm16Impl() override {}

  void TransferSamplesIntoBuffer(const AudioBus* audio_bus,
                                 int source_offset,
                                 int buffer_fill_offset,
                                 int num_samples) override {
    audio_bus->ToInterleavedPartial(
        source_offset,
        num_samples,
        sizeof(int16),
        buffer_.get() + buffer_fill_offset * num_channels_);
  }

  bool EncodeFromFilledBuffer(std::string* out) override {
    // Output 16-bit PCM integers in big-endian byte order.
    out->resize(num_channels_ * samples_per_frame_ * sizeof(int16));
    const int16* src = buffer_.get();
    const int16* const src_end = src + num_channels_ * samples_per_frame_;
    uint16* dest = reinterpret_cast<uint16*>(&out->at(0));
    for (; src < src_end; ++src, ++dest)
      *dest = base::HostToNet16(*src);
    return true;
  }

 private:
  const scoped_ptr<int16[]> buffer_;

  DISALLOW_COPY_AND_ASSIGN(Pcm16Impl);
};

AudioEncoder::AudioEncoder(
    const scoped_refptr<CastEnvironment>& cast_environment,
    int num_channels,
    int sampling_rate,
    int bitrate,
    Codec codec,
    const FrameEncodedCallback& frame_encoded_callback)
    : cast_environment_(cast_environment) {
  // Note: It doesn't matter which thread constructs AudioEncoder, just so long
  // as all calls to InsertAudio() are by the same thread.
  insert_thread_checker_.DetachFromThread();
  switch (codec) {
#if !defined(OS_IOS)
    case CODEC_AUDIO_OPUS:
      impl_ = new OpusImpl(cast_environment,
                           num_channels,
                           sampling_rate,
                           bitrate,
                           frame_encoded_callback);
      break;
#endif
#if defined(OS_MACOSX)
    case CODEC_AUDIO_AAC:
      impl_ = new AppleAacImpl(cast_environment,
                               num_channels,
                               sampling_rate,
                               bitrate,
                               frame_encoded_callback);
      break;
#endif  // defined(OS_MACOSX)
    case CODEC_AUDIO_PCM16:
      impl_ = new Pcm16Impl(cast_environment,
                            num_channels,
                            sampling_rate,
                            frame_encoded_callback);
      break;
    default:
      NOTREACHED() << "Unsupported or unspecified codec for audio encoder";
      break;
  }
}

AudioEncoder::~AudioEncoder() {}

OperationalStatus AudioEncoder::InitializationResult() const {
  DCHECK(insert_thread_checker_.CalledOnValidThread());
  if (impl_.get()) {
    return impl_->InitializationResult();
  }
  return STATUS_UNSUPPORTED_CODEC;
}

int AudioEncoder::GetSamplesPerFrame() const {
  DCHECK(insert_thread_checker_.CalledOnValidThread());
  if (InitializationResult() != STATUS_INITIALIZED) {
    NOTREACHED();
    return std::numeric_limits<int>::max();
  }
  return impl_->samples_per_frame();
}

base::TimeDelta AudioEncoder::GetFrameDuration() const {
  DCHECK(insert_thread_checker_.CalledOnValidThread());
  if (InitializationResult() != STATUS_INITIALIZED) {
    NOTREACHED();
    return base::TimeDelta();
  }
  return impl_->frame_duration();
}

void AudioEncoder::InsertAudio(scoped_ptr<AudioBus> audio_bus,
                               const base::TimeTicks& recorded_time) {
  DCHECK(insert_thread_checker_.CalledOnValidThread());
  DCHECK(audio_bus.get());
  if (InitializationResult() != STATUS_INITIALIZED) {
    NOTREACHED();
    return;
  }
  cast_environment_->PostTask(CastEnvironment::AUDIO,
                              FROM_HERE,
                              base::Bind(&AudioEncoder::ImplBase::EncodeAudio,
                                         impl_,
                                         base::Passed(&audio_bus),
                                         recorded_time));
}

}  // namespace cast
}  // namespace media