Add FFmpeg audio decoder for the clear key CDM.

webkit/media/crypto/ppapi/ffmpeg_cdm_audio_decoder.cc

+// 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 "webkit/media/crypto/ppapi/ffmpeg_cdm_audio_decoder.h"
+
+#include <algorithm>
+
+#include "base/logging.h"
+#include "media/base/buffers.h"
+#include "media/base/limits.h"
+#include "webkit/media/crypto/ppapi/content_decryption_module.h"
+
+// Include FFmpeg header files.
+extern "C" {
+// Temporarily disable possible loss of data warning.
+MSVC_PUSH_DISABLE_WARNING(4244);
+#include <libavcodec/avcodec.h>
+MSVC_POP_WARNING();
+}  // extern "C"
+
+namespace webkit_media {
+
+// Maximum number of channels with defined order in the Vorbis specification.
+// http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html
+static const int kMaxVorbisChannels = 8;
+
+static CodecID CdmAudioCodecToCodecID(
+    cdm::AudioDecoderConfig::AudioCodec audio_codec) {
+  switch (audio_codec) {
+    case cdm::AudioDecoderConfig::kCodecVorbis:
+      return CODEC_ID_VORBIS;
+    default:
+      NOTREACHED() << "Unsupported cdm::AudioCodec: " << audio_codec;
+  }
+
+  return CODEC_ID_NONE;
+}
+
+static void CdmAudioDecoderConfigToAVCodecContext(
+    const cdm::AudioDecoderConfig& config,
+    AVCodecContext* codec_context) {
+  codec_context->codec_type = AVMEDIA_TYPE_AUDIO;
+  codec_context->codec_id = CdmAudioCodecToCodecID(config.codec);
+
+  switch (config.bits_per_channel) {
+    case 8:
+      codec_context->sample_fmt = AV_SAMPLE_FMT_U8;
+      break;
+    case 16:
+      codec_context->sample_fmt = AV_SAMPLE_FMT_S16;
+      break;
+    case 32:
+      codec_context->sample_fmt = AV_SAMPLE_FMT_S32;
+      break;
+    default:
+      DVLOG(1) << "CdmAudioDecoderConfigToAVCodecContext() Unsupported bits "
+                  "per channel: " << config.bits_per_channel;
+      codec_context->sample_fmt = AV_SAMPLE_FMT_NONE;
+  }
+
+  codec_context->channels = config.channel_count;
+  codec_context->sample_rate = config.samples_per_second;
+
+  if (config.extra_data) {
+    codec_context->extradata_size = config.extra_data_size;
+    codec_context->extradata = reinterpret_cast<uint8_t*>(
+        av_malloc(config.extra_data_size + FF_INPUT_BUFFER_PADDING_SIZE));
+    memcpy(codec_context->extradata, config.extra_data,
+           config.extra_data_size);
+    memset(codec_context->extradata + config.extra_data_size, '\0',
+           FF_INPUT_BUFFER_PADDING_SIZE);
+  } else {
+    codec_context->extradata = NULL;
+    codec_context->extradata_size = 0;
+  }
+}
+
+// Returns true when the decode result was end of stream.
+static inline bool IsEndOfOutputStream(int result,
+                                       int decoded_size,
+                                       bool is_end_of_input_stream) {
+  // 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 && is_end_of_input_stream;
+}
+
+FFmpegCdmAudioDecoder::FFmpegCdmAudioDecoder(cdm::Allocator* allocator)
+    : is_initialized_(false),
+      allocator_(allocator),
+      codec_context_(NULL),
+      av_frame_(NULL),
+      bits_per_channel_(0),
+      samples_per_second_(0),
+      bytes_per_frame_(0),
+      output_timestamp_base_(media::kNoTimestamp()),
+      total_frames_decoded_(0),
+      last_input_timestamp_(media::kNoTimestamp()),
+      output_bytes_to_drop_(0) {
+}
+
+FFmpegCdmAudioDecoder::~FFmpegCdmAudioDecoder() {
+  ReleaseFFmpegResources();
+}
+
+bool FFmpegCdmAudioDecoder::Initialize(const cdm::AudioDecoderConfig& config) {
+  DVLOG(1) << "Initialize()";
+
+  if (!IsValidConfig(config)) {
+    LOG(ERROR) << "Initialize(): invalid audio decoder configuration.";
+    return false;
+  }
+
+  if (is_initialized_) {
+    LOG(ERROR) << "Initialize(): Already initialized.";
+    return false;
+  }
+
+  // Initialize AVCodecContext structure.
+  codec_context_ = avcodec_alloc_context3(NULL);
+  CdmAudioDecoderConfigToAVCodecContext(config, codec_context_);
+  DCHECK_EQ(CODEC_ID_VORBIS, codec_context_->codec_id);
+
+  AVCodec* codec = avcodec_find_decoder(codec_context_->codec_id);
+  if (!codec) {
+    LOG(ERROR) << "Initialize(): avcodec_find_decoder failed.";
+    return false;
+  }
+
+  int status;
+  if ((status = avcodec_open2(codec_context_, codec, NULL)) < 0) {
+    LOG(ERROR) << "Initialize(): avcodec_open2 failed: " << status;
+    return false;
+  }
+
+  av_frame_ = avcodec_alloc_frame();
+  bits_per_channel_ = config.bits_per_channel;
+  samples_per_second_ = config.samples_per_second;
+  bytes_per_frame_ = codec_context_->channels * bits_per_channel_ / 8;
+  serialized_audio_frames_.reserve(bytes_per_frame_ * samples_per_second_);
+  is_initialized_ = true;
+
+  return true;
+}
+
+void FFmpegCdmAudioDecoder::Deinitialize() {
+  DVLOG(1) << "Deinitialize()";
+  ReleaseFFmpegResources();
+  is_initialized_ = false;
+  ResetAudioTimingData();
+}
+
+void FFmpegCdmAudioDecoder::Reset() {
+  DVLOG(1) << "Reset()";
+  avcodec_flush_buffers(codec_context_);
+  ResetAudioTimingData();
+}
+
+// static
+bool FFmpegCdmAudioDecoder::IsValidConfig(
+    const cdm::AudioDecoderConfig& config) {
+  return config.codec == cdm::AudioDecoderConfig::kCodecVorbis &&
+         config.channel_count > 0 &&
+         config.channel_count <= kMaxVorbisChannels &&
+         config.bits_per_channel > 0 &&
+         config.bits_per_channel <= media::limits::kMaxBitsPerSample &&
+         config.samples_per_second > 0 &&
+         config.samples_per_second <= media::limits::kMaxSampleRate;
+}
+
+cdm::Status FFmpegCdmAudioDecoder::DecodeBuffer(
+    const uint8_t* compressed_buffer,
+    int32_t compressed_buffer_size,
+    int64_t input_timestamp,
+    cdm::AudioFrames* decoded_frames) {
+  const bool is_end_of_stream = compressed_buffer_size == 0;
+  base::TimeDelta timestamp =
+      base::TimeDelta::FromMicroseconds(input_timestamp);
+  if (!is_end_of_stream) {
+    if (last_input_timestamp_ == media::kNoTimestamp()) {
+      if (codec_context_->codec_id == CODEC_ID_VORBIS &&
+          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
+        int frames_to_drop = floor(
+            0.5 + -timestamp.InSecondsF() * samples_per_second_);
+        output_bytes_to_drop_ = bytes_per_frame_ * frames_to_drop;
+      } else {
+        last_input_timestamp_ = timestamp;
+      }
+    } else if (timestamp != media::kNoTimestamp()) {
+      if (timestamp < last_input_timestamp_) {
+        base::TimeDelta diff = timestamp - last_input_timestamp_;
+        DVLOG(1) << "Input timestamps are not monotonically increasing! "
+                 << " ts " << timestamp.InMicroseconds() << " us"
+                 << " diff " << diff.InMicroseconds() << " us";
+        return cdm::kDecodeError;
+      }
+
+      last_input_timestamp_ = timestamp;
+    }
+  }
+
+  AVPacket packet;
+  av_init_packet(&packet);
+  packet.data = const_cast<uint8_t*>(compressed_buffer);
+  packet.size = compressed_buffer_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 {
+    // Reset frame to default values.
+    avcodec_get_frame_defaults(av_frame_);
+
+    int frame_decoded = 0;
+    int result = avcodec_decode_audio4(
+        codec_context_, av_frame_, &frame_decoded, &packet);
+
+    if (result < 0) {
+      DCHECK(!is_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.";
+
+      DLOG(ERROR)
+          << "Error decoding an audio frame with timestamp: "
+          << timestamp.InMicroseconds() << " us, duration: "
+          << timestamp.InMicroseconds() << " us, packet size: "
+          << compressed_buffer_size << " bytes";
+
+      return cdm::kDecodeError;
+    }
+
+    // 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;
+
+    if (output_timestamp_base_ == media::kNoTimestamp() && !is_end_of_stream) {
+      DCHECK(timestamp != media::kNoTimestamp());
+      if (output_bytes_to_drop_ > 0) {
+        // If we have to drop samples it always means the timeline starts at 0.
+        output_timestamp_base_ = base::TimeDelta();
+      } else {
+        output_timestamp_base_ = timestamp;
+      }
+    }
+
+    const uint8_t* decoded_audio_data = NULL;
+    int decoded_audio_size = 0;
+    if (frame_decoded) {
+      int output_sample_rate = av_frame_->sample_rate;
+      if (output_sample_rate != samples_per_second_) {
+        DLOG(ERROR) << "Output sample rate (" << output_sample_rate
+                    << ") doesn't match expected rate " << samples_per_second_;
+        return cdm::kDecodeError;
+      }
+
+      decoded_audio_data = av_frame_->data[0];
+      decoded_audio_size = av_samples_get_buffer_size(
+          NULL, codec_context_->channels, av_frame_->nb_samples,
+          codec_context_->sample_fmt, 1);
+    }
+
+    if (decoded_audio_size > 0 && output_bytes_to_drop_ > 0) {
+      int dropped_size = std::min(decoded_audio_size, output_bytes_to_drop_);
+      decoded_audio_data += dropped_size;
+      decoded_audio_size -= dropped_size;
+      output_bytes_to_drop_ -= dropped_size;
+    }
+
+    if (decoded_audio_size > 0) {
+      DCHECK_EQ(decoded_audio_size % bytes_per_frame_, 0)
+          << "Decoder didn't output full frames";
+
+      base::TimeDelta output_timestamp = GetNextOutputTimestamp();
+      total_frames_decoded_ += decoded_audio_size / bytes_per_frame_;
+
+      // Serialize the audio samples into |serialized_audio_frames_|.
+      SerializeInt64(output_timestamp.InMicroseconds());
+      SerializeInt64(decoded_audio_size);
+      serialized_audio_frames_.insert(
+          serialized_audio_frames_.end(),
+          decoded_audio_data,
+          decoded_audio_data + decoded_audio_size);
+    } else if (IsEndOfOutputStream(result,
+                                   decoded_audio_size,
+                                   is_end_of_stream)) {
+      DCHECK_EQ(packet.size, 0);
+      return cdm::kNeedMoreData;
+    }
+  } while (packet.size > 0);
+
+  if (serialized_audio_frames_.size() > 0) {
+    decoded_frames->set_buffer(
+        allocator_->Allocate(serialized_audio_frames_.size()));
+    if (!decoded_frames->buffer()) {
+      LOG(ERROR) << "DecodeBuffer() cdm::Allocator::Allocate failed.";
+      return cdm::kDecodeError;
+    }
+    memcpy(decoded_frames->buffer()->data(),
+           &serialized_audio_frames_[0],
+           serialized_audio_frames_.size());
+    serialized_audio_frames_.clear();
+  }
+
+  return cdm::kSuccess;
+}
+
+void FFmpegCdmAudioDecoder::ResetAudioTimingData() {
+  output_timestamp_base_ = media::kNoTimestamp();
+  total_frames_decoded_ = 0;
+  last_input_timestamp_ = media::kNoTimestamp();
+  output_bytes_to_drop_ = 0;
+}
+
+void FFmpegCdmAudioDecoder::ReleaseFFmpegResources() {
+  DVLOG(1) << "ReleaseFFmpegResources()";
+
+  if (codec_context_) {
+    av_free(codec_context_->extradata);
+    avcodec_close(codec_context_);
+    av_free(codec_context_);
+    codec_context_ = NULL;
+  }
+  if (av_frame_) {
+    av_free(av_frame_);
+    av_frame_ = NULL;
+  }
+}
+
+base::TimeDelta FFmpegCdmAudioDecoder::GetNextOutputTimestamp() const {
+  DCHECK(output_timestamp_base_ != media::kNoTimestamp());
+  double decoded_us = (total_frames_decoded_ / samples_per_second_) *

[xhwang, 2012/10/24 22:24:19]

samples_per_second can be large, e.g. 44100. We could lose a lot of precision
here. Do the multiplication first, or cast to double before doing division?

[Tom Finegan, 2012/10/24 23:37:52]

Done. Are the consts OK?

+      base::Time::kMicrosecondsPerSecond;
+  return output_timestamp_base_ +
+      base::TimeDelta::FromMicroseconds(decoded_us);
+}
+
+void FFmpegCdmAudioDecoder::SerializeInt64(int64 value) {
+  const uint8_t* ptr = reinterpret_cast<uint8_t*>(&value);
+  serialized_audio_frames_.insert(serialized_audio_frames_.end(),
+                                  ptr, ptr + sizeof(value));
+}
+
+}  // namespace webkit_media