// 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 <stdint.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/timestamp_constants.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_t* 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, s->codec_id);
  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);
  }

  if (s->sample_rate != frame->sample_rate) {
    DLOG(ERROR) << "AVCodecContext and AVFrame disagree on sample rate."
                << s->sample_rate << " vs " << frame->sample_rate;
    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_t**>(
        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 scoped_refptr<MediaLog>& media_log)
    : task_runner_(task_runner),
      state_(kUninitialized),
      av_sample_format_(0),
      media_log_(media_log) {
}

FFmpegAudioDecoder::~FFmpegAudioDecoder() {
  DCHECK(task_runner_->BelongsToCurrentThread());

  if (state_ != kUninitialized)
    ReleaseFFmpegResources();
}

std::string FFmpegAudioDecoder::GetDisplayName() const {
  return "FFmpegAudioDecoder";
}

void FFmpegAudioDecoder::Initialize(const AudioDecoderConfig& config,
                                    CdmContext* /* cdm_context */,
                                    const InitCB& init_cb,
                                    const OutputCB& output_cb) {
  DCHECK(task_runner_->BelongsToCurrentThread());
  DCHECK(config.IsValidConfig());

  InitCB bound_init_cb = BindToCurrentLoop(init_cb);

  if (config.is_encrypted()) {
    bound_init_cb.Run(false);
    return;
  }

  FFmpegGlue::InitializeFFmpeg();
  config_ = config;

  // TODO(xhwang): Only set |config_| after we successfully configure the
  // decoder. Make sure we clean up all member variables upon failure.
  if (!ConfigureDecoder()) {
    bound_init_cb.Run(false);
    return;
  }

  // Success!
  output_cb_ = BindToCurrentLoop(output_cb);
  state_ = kNormal;
  bound_init_cb.Run(true);
}

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(DecodeStatus::DECODE_ERROR);
    return;
  }

  // Do nothing if decoding has finished.
  if (state_ == kDecodeFinished) {
    decode_cb_bound.Run(DecodeStatus::OK);
    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::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.get());

  // 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(DecodeStatus::DECODE_ERROR);
    return;
  }

  bool has_produced_frame;
  do {
    has_produced_frame = false;
    if (!FFmpegDecode(buffer, &has_produced_frame)) {
      state_ = kError;
      decode_cb.Run(DecodeStatus::DECODE_ERROR);
      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(DecodeStatus::OK);
}

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_t*>(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(DEBUG, media_log_)
          << "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;

    bool config_changed = false;
    if (frame_decoded) {
      const int channels = DetermineChannels(av_frame_.get());
      ChannelLayout channel_layout = ChannelLayoutToChromeChannelLayout(
          codec_context_->channel_layout, codec_context_->channels);

      bool is_sample_rate_change =
          av_frame_->sample_rate != config_.samples_per_second();
      bool is_config_stale =
          is_sample_rate_change ||
          channels != ChannelLayoutToChannelCount(config_.channel_layout()) ||
          av_frame_->format != av_sample_format_;

      // Only consider channel layout changes for AAC.
      // TODO(tguilbert, dalecurtis): Due to http://crbug.com/600538 we need to
      // allow channel layout changes for the moment. See if ffmpeg is fixable.
      if (config_.codec() == kCodecAAC)
        is_config_stale |= channel_layout != config_.channel_layout();

      if (is_config_stale) {
        // Only allow midstream configuration changes for AAC. Sample format is
        // not expected to change between AAC profiles.
        if (config_.codec() == kCodecAAC &&
            av_frame_->format == av_sample_format_) {
          MEDIA_LOG(DEBUG, media_log_)
              << " Detected AAC midstream configuration change"
              << " PTS:" << buffer->timestamp().InMicroseconds()
              << " Sample Rate: " << av_frame_->sample_rate << " vs "
              << config_.samples_per_second()
              << ", ChannelLayout: " << channel_layout << " vs "
              << config_.channel_layout() << ", Channels: " << channels
              << " vs "
              << ChannelLayoutToChannelCount(config_.channel_layout());
          config_.Initialize(config_.codec(), config_.sample_format(),
                             channel_layout, av_frame_->sample_rate,
                             config_.extra_data(), config_.encryption_scheme(),
                             config_.seek_preroll(), config_.codec_delay());
          config_changed = true;
          if (is_sample_rate_change)
            ResetTimestampState();
        } else {
          MEDIA_LOG(ERROR, media_log_)
              << "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_;
          // 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)) {
      if (config_changed &&
          output->sample_rate() != config_.samples_per_second()) {
        // At the boundary of the config change, FFmpeg's AAC decoder gives the
        // previous sample rate when calling our GetAudioBuffer. Set the correct
        // sample rate before sending the buffer along.
        // TODO(chcunningham): Fix FFmpeg and upstream it.
        output->AdjustSampleRate(config_.samples_per_second());
      }
      *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() {
  DCHECK(config_.IsValidConfig());
  DCHECK(!config_.is_encrypted());

  // 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());
  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(new AudioDiscardHelper(config_.samples_per_second(),
                                               config_.codec_delay()));
  discard_helper_->Reset(config_.codec_delay());
}

}  // namespace media