// 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.

#include "media/filters/ffmpeg_audio_decoder.h"

#include "base/callback_helpers.h"
#include "base/single_thread_task_runner.h"
#include "media/base/audio_buffer.h"
#include "media/base/audio_bus.h"
#include "media/base/audio_decoder_config.h"
#include "media/base/audio_discard_helper.h"
#include "media/base/bind_to_current_loop.h"
#include "media/base/decoder_buffer.h"
#include "media/base/limits.h"
#include "media/base/sample_format.h"
#include "media/ffmpeg/ffmpeg_common.h"
#include "media/filters/ffmpeg_glue.h"

namespace media {

// Returns true if the decode result was end of stream.
static inline bool IsEndOfStream(int result,
                                 int decoded_size,
                                 const scoped_refptr<DecoderBuffer>& input) {
  // Three conditions to meet to declare end of stream for this decoder:
  // 1. FFmpeg didn't read anything.
  // 2. FFmpeg didn't output anything.
  // 3. An end of stream buffer is received.
  return result == 0 && decoded_size == 0 && input->end_of_stream();
}

// Return the number of channels from the data in |frame|.
static inline int DetermineChannels(AVFrame* frame) {
#if defined(CHROMIUM_NO_AVFRAME_CHANNELS)
  // When use_system_ffmpeg==1, libav's AVFrame doesn't have channels field.
  return av_get_channel_layout_nb_channels(frame->channel_layout);
#else
  return frame->channels;
#endif
}

// Called by FFmpeg's allocation routine to free a buffer. |opaque| is the
// AudioBuffer allocated, so unref it.
static void ReleaseAudioBufferImpl(void* opaque, uint8* data) {
  scoped_refptr<AudioBuffer> buffer;
  buffer.swap(reinterpret_cast<AudioBuffer**>(&opaque));
}

// Called by FFmpeg's allocation routine to allocate a buffer. Uses
// AVCodecContext.opaque to get the object reference in order to call
// GetAudioBuffer() to do the actual allocation.
static int GetAudioBuffer(struct AVCodecContext* s, AVFrame* frame, int flags) {
  DCHECK(s->codec->capabilities & CODEC_CAP_DR1);
  DCHECK_EQ(s->codec_type, AVMEDIA_TYPE_AUDIO);

  // Since this routine is called by FFmpeg when a buffer is required for audio
  // data, use the values supplied by FFmpeg (ignoring the current settings).
  // FFmpegDecode() gets to determine if the buffer is useable or not.
  AVSampleFormat format = static_cast<AVSampleFormat>(frame->format);
  SampleFormat sample_format = AVSampleFormatToSampleFormat(format);
  int channels = DetermineChannels(frame);
  if (channels <= 0 || channels >= limits::kMaxChannels) {
    DLOG(ERROR) << "Requested number of channels (" << channels
                << ") exceeds limit.";
    return AVERROR(EINVAL);
  }

  int bytes_per_channel = SampleFormatToBytesPerChannel(sample_format);
  if (frame->nb_samples <= 0)
    return AVERROR(EINVAL);

  if (s->channels != channels) {
    DLOG(ERROR) << "AVCodecContext and AVFrame disagree on channel count.";
    return AVERROR(EINVAL);
  }

  // Determine how big the buffer should be and allocate it. FFmpeg may adjust
  // how big each channel data is in order to meet the alignment policy, so
  // we need to take this into consideration.
  int buffer_size_in_bytes =
      av_samples_get_buffer_size(&frame->linesize[0],
                                 channels,
                                 frame->nb_samples,
                                 format,
                                 AudioBuffer::kChannelAlignment);
  // Check for errors from av_samples_get_buffer_size().
  if (buffer_size_in_bytes < 0)
    return buffer_size_in_bytes;
  int frames_required = buffer_size_in_bytes / bytes_per_channel / channels;
  DCHECK_GE(frames_required, frame->nb_samples);
  scoped_refptr<AudioBuffer> buffer = AudioBuffer::CreateBuffer(
      sample_format,
      ChannelLayoutToChromeChannelLayout(s->channel_layout, s->channels),
      channels,
      s->sample_rate,
      frames_required);

  // Initialize the data[] and extended_data[] fields to point into the memory
  // allocated for AudioBuffer. |number_of_planes| will be 1 for interleaved
  // audio and equal to |channels| for planar audio.
  int number_of_planes = buffer->channel_data().size();
  if (number_of_planes <= AV_NUM_DATA_POINTERS) {
    DCHECK_EQ(frame->extended_data, frame->data);
    for (int i = 0; i < number_of_planes; ++i)
      frame->data[i] = buffer->channel_data()[i];
  } else {
    // There are more channels than can fit into data[], so allocate
    // extended_data[] and fill appropriately.
    frame->extended_data = static_cast<uint8**>(
        av_malloc(number_of_planes * sizeof(*frame->extended_data)));
    int i = 0;
    for (; i < AV_NUM_DATA_POINTERS; ++i)
      frame->extended_data[i] = frame->data[i] = buffer->channel_data()[i];
    for (; i < number_of_planes; ++i)
      frame->extended_data[i] = buffer->channel_data()[i];
  }

  // Now create an AVBufferRef for the data just allocated. It will own the
  // reference to the AudioBuffer object.
  void* opaque = NULL;
  buffer.swap(reinterpret_cast<AudioBuffer**>(&opaque));
  frame->buf[0] = av_buffer_create(
      frame->data[0], buffer_size_in_bytes, ReleaseAudioBufferImpl, opaque, 0);
  return 0;
}

FFmpegAudioDecoder::FFmpegAudioDecoder(
    const scoped_refptr<base::SingleThreadTaskRunner>& task_runner,
    const LogCB& log_cb)
    : task_runner_(task_runner),
      state_(kUninitialized),
      av_sample_format_(0),
      log_cb_(log_cb) {
}

FFmpegAudioDecoder::~FFmpegAudioDecoder() {
  DCHECK_EQ(state_, kUninitialized);
  DCHECK(!codec_context_);
  DCHECK(!av_frame_);
}

void FFmpegAudioDecoder::Initialize(const AudioDecoderConfig& config,
                                    const PipelineStatusCB& status_cb,
                                    const OutputCB& output_cb) {
  DCHECK(task_runner_->BelongsToCurrentThread());
  DCHECK(!config.is_encrypted());

  FFmpegGlue::InitializeFFmpeg();

  config_ = config;
  PipelineStatusCB initialize_cb = BindToCurrentLoop(status_cb);

  if (!config.IsValidConfig() || !ConfigureDecoder()) {
    initialize_cb.Run(DECODER_ERROR_NOT_SUPPORTED);
    return;
  }

  // Success!
  output_cb_ = BindToCurrentLoop(output_cb);
  state_ = kNormal;
  initialize_cb.Run(PIPELINE_OK);
}

void FFmpegAudioDecoder::Decode(const scoped_refptr<DecoderBuffer>& buffer,
                                const DecodeCB& decode_cb) {
  DCHECK(task_runner_->BelongsToCurrentThread());
  DCHECK(!decode_cb.is_null());
  CHECK_NE(state_, kUninitialized);
  DecodeCB decode_cb_bound = BindToCurrentLoop(decode_cb);

  if (state_ == kError) {
    decode_cb_bound.Run(kDecodeError);
    return;
  }

  // Do nothing if decoding has finished.
  if (state_ == kDecodeFinished) {
    decode_cb_bound.Run(kOk);
    return;
  }

  DecodeBuffer(buffer, decode_cb_bound);
}

void FFmpegAudioDecoder::Reset(const base::Closure& closure) {
  DCHECK(task_runner_->BelongsToCurrentThread());

  avcodec_flush_buffers(codec_context_.get());
  state_ = kNormal;
  ResetTimestampState();
  task_runner_->PostTask(FROM_HERE, closure);
}

void FFmpegAudioDecoder::Stop() {
  DCHECK(task_runner_->BelongsToCurrentThread());

  if (state_ == kUninitialized)
    return;

  ReleaseFFmpegResources();
  ResetTimestampState();
  state_ = kUninitialized;
}

void FFmpegAudioDecoder::DecodeBuffer(
    const scoped_refptr<DecoderBuffer>& buffer,
    const DecodeCB& decode_cb) {
  DCHECK(task_runner_->BelongsToCurrentThread());
  DCHECK_NE(state_, kUninitialized);
  DCHECK_NE(state_, kDecodeFinished);
  DCHECK_NE(state_, kError);

  DCHECK(buffer);

  // Make sure we are notified if http://crbug.com/49709 returns.  Issue also
  // occurs with some damaged files.
  if (!buffer->end_of_stream() && buffer->timestamp() == kNoTimestamp()) {
    DVLOG(1) << "Received a buffer without timestamps!";
    decode_cb.Run(kDecodeError);
    return;
  }

  if (!buffer->end_of_stream() && !discard_helper_->initialized() &&
      codec_context_->codec_id == AV_CODEC_ID_VORBIS &&
      buffer->timestamp() < base::TimeDelta()) {
    // Dropping frames for negative timestamps as outlined in section A.2
    // in the Vorbis spec. http://xiph.org/vorbis/doc/Vorbis_I_spec.html
    const int discard_frames =
        discard_helper_->TimeDeltaToFrames(-buffer->timestamp());
    discard_helper_->Reset(discard_frames);
  }

  bool has_produced_frame;
  do {
    has_produced_frame = false;
    if (!FFmpegDecode(buffer, &has_produced_frame)) {
      state_ = kError;
      decode_cb.Run(kDecodeError);
      return;
    }
    // Repeat to flush the decoder after receiving EOS buffer.
  } while (buffer->end_of_stream() && has_produced_frame);

  if (buffer->end_of_stream())
    state_ = kDecodeFinished;

  decode_cb.Run(kOk);
}

bool FFmpegAudioDecoder::FFmpegDecode(
    const scoped_refptr<DecoderBuffer>& buffer,
    bool* has_produced_frame) {
  DCHECK(!*has_produced_frame);

  AVPacket packet;
  av_init_packet(&packet);
  if (buffer->end_of_stream()) {
    packet.data = NULL;
    packet.size = 0;
  } else {
    packet.data = const_cast<uint8*>(buffer->data());
    packet.size = buffer->data_size();
  }

  // Each audio packet may contain several frames, so we must call the decoder
  // until we've exhausted the packet.  Regardless of the packet size we always
  // want to hand it to the decoder at least once, otherwise we would end up
  // skipping end of stream packets since they have a size of zero.
  do {
    int frame_decoded = 0;
    const int result = avcodec_decode_audio4(
        codec_context_.get(), av_frame_.get(), &frame_decoded, &packet);

    if (result < 0) {
      DCHECK(!buffer->end_of_stream())
          << "End of stream buffer produced an error! "
          << "This is quite possibly a bug in the audio decoder not handling "
          << "end of stream AVPackets correctly.";

      MEDIA_LOG(log_cb_)
          << "Dropping audio frame which failed decode with timestamp: "
          << buffer->timestamp().InMicroseconds() << " us, duration: "
          << buffer->duration().InMicroseconds() << " us, packet size: "
          << buffer->data_size() << " bytes";

      break;
    }

    // Update packet size and data pointer in case we need to call the decoder
    // with the remaining bytes from this packet.
    packet.size -= result;
    packet.data += result;

    scoped_refptr<AudioBuffer> output;
    const int channels = DetermineChannels(av_frame_.get());
    if (frame_decoded) {
      if (av_frame_->sample_rate != config_.samples_per_second() ||
          channels != ChannelLayoutToChannelCount(config_.channel_layout()) ||
          av_frame_->format != av_sample_format_) {
        DLOG(ERROR) << "Unsupported midstream configuration change!"
                    << " Sample Rate: " << av_frame_->sample_rate << " vs "
                    << config_.samples_per_second()
                    << ", Channels: " << channels << " vs "
                    << ChannelLayoutToChannelCount(config_.channel_layout())
                    << ", Sample Format: " << av_frame_->format << " vs "
                    << av_sample_format_;

        if (config_.codec() == kCodecAAC &&
            av_frame_->sample_rate == 2 * config_.samples_per_second()) {
          MEDIA_LOG(log_cb_) << "Implicit HE-AAC signalling is being used."
                             << " Please use mp4a.40.5 instead of mp4a.40.2 in"
                             << " the mimetype.";
        }
        // This is an unrecoverable error, so bail out.
        av_frame_unref(av_frame_.get());
        return false;
      }

      // Get the AudioBuffer that the data was decoded into. Adjust the number
      // of frames, in case fewer than requested were actually decoded.
      output = reinterpret_cast<AudioBuffer*>(
          av_buffer_get_opaque(av_frame_->buf[0]));

      DCHECK_EQ(ChannelLayoutToChannelCount(config_.channel_layout()),
                output->channel_count());
      const int unread_frames = output->frame_count() - av_frame_->nb_samples;
      DCHECK_GE(unread_frames, 0);
      if (unread_frames > 0)
        output->TrimEnd(unread_frames);
      av_frame_unref(av_frame_.get());
    }

    // WARNING: |av_frame_| no longer has valid data at this point.
    const int decoded_frames = frame_decoded ? output->frame_count() : 0;
    if (IsEndOfStream(result, decoded_frames, buffer)) {
      DCHECK_EQ(packet.size, 0);
    } else if (discard_helper_->ProcessBuffers(buffer, output)) {
      *has_produced_frame = true;
      output_cb_.Run(output);
    }
  } while (packet.size > 0);

  return true;
}

void FFmpegAudioDecoder::ReleaseFFmpegResources() {
  codec_context_.reset();
  av_frame_.reset();
}

bool FFmpegAudioDecoder::ConfigureDecoder() {
  if (!config_.IsValidConfig()) {
    DLOG(ERROR) << "Invalid audio stream -"
                << " codec: " << config_.codec()
                << " channel layout: " << config_.channel_layout()
                << " bits per channel: " << config_.bits_per_channel()
                << " samples per second: " << config_.samples_per_second();
    return false;
  }

  if (config_.is_encrypted()) {
    DLOG(ERROR) << "Encrypted audio stream not supported";
    return false;
  }

  // Release existing decoder resources if necessary.
  ReleaseFFmpegResources();

  // Initialize AVCodecContext structure.
  codec_context_.reset(avcodec_alloc_context3(NULL));
  AudioDecoderConfigToAVCodecContext(config_, codec_context_.get());

  codec_context_->opaque = this;
  codec_context_->get_buffer2 = GetAudioBuffer;
  codec_context_->refcounted_frames = 1;

  AVCodec* codec = avcodec_find_decoder(codec_context_->codec_id);
  if (!codec || avcodec_open2(codec_context_.get(), codec, NULL) < 0) {
    DLOG(ERROR) << "Could not initialize audio decoder: "
                << codec_context_->codec_id;
    ReleaseFFmpegResources();
    state_ = kUninitialized;
    return false;
  }

  // Success!
  av_frame_.reset(av_frame_alloc());
  discard_helper_.reset(new AudioDiscardHelper(config_.samples_per_second(),
                                               config_.codec_delay()));
  av_sample_format_ = codec_context_->sample_fmt;

  if (codec_context_->channels !=
      ChannelLayoutToChannelCount(config_.channel_layout())) {
    DLOG(ERROR) << "Audio configuration specified "
                << ChannelLayoutToChannelCount(config_.channel_layout())
                << " channels, but FFmpeg thinks the file contains "
                << codec_context_->channels << " channels";
    ReleaseFFmpegResources();
    state_ = kUninitialized;
    return false;
  }

  ResetTimestampState();
  return true;
}

void FFmpegAudioDecoder::ResetTimestampState() {
  discard_helper_->Reset(config_.codec_delay());
}

}  // namespace media