Consolidate and improve audio decoding test for all decoders.

media/filters/audio_decoder_unittest.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 <deque>
+
+#include "base/bind.h"
+#include "base/format_macros.h"
+#include "base/md5.h"
+#include "base/message_loop/message_loop.h"
+#include "base/run_loop.h"
+#include "base/strings/stringprintf.h"
+#include "build/build_config.h"
+#include "media/base/audio_buffer.h"
+#include "media/base/audio_bus.h"
+#include "media/base/audio_hash.h"
+#include "media/base/decoder_buffer.h"
+#include "media/base/test_data_util.h"
+#include "media/base/test_helpers.h"
+#include "media/ffmpeg/ffmpeg_common.h"
+#include "media/filters/audio_file_reader.h"
+#include "media/filters/ffmpeg_audio_decoder.h"
+#include "media/filters/in_memory_url_protocol.h"
+#include "media/filters/opus_audio_decoder.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+namespace media {
+
+// The number of packets to read and then decode from each file.
+static const size_t kDecodeRuns = 3;
+
+enum AudioDecoderType {
+  FFMPEG,
+  OPUS,
+};
+
+struct DecodedBufferExpectations {
+  const int64 timestamp;
+  const int64 duration;
+  const char* hash;
+};
+
+struct DecoderTestData {
+  const AudioDecoderType decoder_type;
+  const AudioCodec codec;
+  const char* filename;
+  const DecodedBufferExpectations* expectations;
+  const int first_packet_pts_us;
+  const int samples_per_second;
+  const ChannelLayout channel_layout;
+};
+
+// Tells gtest how to print our DecoderTestData structure.
+std::ostream& operator<<(std::ostream& os, const DecoderTestData& data) {
+  return os << data.filename;
+}
+
+class AudioDecoderTest : public testing::TestWithParam<DecoderTestData> {
+ public:
+  AudioDecoderTest() : pending_decode_(false), pending_reset_(false) {
+    switch (GetParam().decoder_type) {
+      case FFMPEG:
+        decoder_.reset(new FFmpegAudioDecoder(
+            message_loop_.message_loop_proxy(), LogCB()));
+        break;
+      case OPUS:
+        decoder_.reset(
+            new OpusAudioDecoder(message_loop_.message_loop_proxy()));
+        break;
+    }
+  }
+
+  virtual ~AudioDecoderTest() {
+    EXPECT_FALSE(pending_decode_);
+    EXPECT_FALSE(pending_reset_);
+  }
+
+ protected:
+  void SendAbort() {
+    ASSERT_FALSE(pending_decode_);
+    pending_decode_ = true;
+    decoder_->Decode(
+        NULL,
+        base::Bind(&AudioDecoderTest::DecodeFinished, base::Unretained(this)));
+    base::RunLoop().RunUntilIdle();
+  }
+
+  void SendEndOfStream() {
+    ASSERT_FALSE(pending_decode_);
+    pending_decode_ = true;
+    decoder_->Decode(
+        DecoderBuffer::CreateEOSBuffer(),
+        base::Bind(&AudioDecoderTest::DecodeFinished, base::Unretained(this)));
+    base::RunLoop().RunUntilIdle();
+  }
+
+  void Initialize() {
+    // Load the test data file.
+    data_ = ReadTestDataFile(GetParam().filename);
+    protocol_.reset(
+        new InMemoryUrlProtocol(data_->data(), data_->data_size(), false));
+    reader_.reset(new AudioFileReader(protocol_.get()));
+    reader_->Open();
+
+    // Load the first packet and check its timestamp.
+    AVPacket packet;
+    ASSERT_TRUE(reader_->ReadPacketForTesting(&packet));
+    EXPECT_EQ(GetParam().first_packet_pts_us, packet.pts);

[wolenetz, 2014/06/16 23:36:08]

Explicitly compare a converted-to-microseconds packet.pts, or rename
first_packet_pts_us to just first_packet_pts. I prefer the former.

[DaleCurtis, 2014/06/19 01:05:52]

Prefer the latter.

+    start_timestamp_ = ConvertFromTimeBase(
+        reader_->codec_context_for_testing()->time_base, packet.pts);
+    av_free_packet(&packet);
+
+    // Seek back to the beginning.
+    ASSERT_TRUE(reader_->SeekForTesting(start_timestamp_));
+
+    AudioDecoderConfig config;
+    AVCodecContextToAudioDecoderConfig(
+        reader_->codec_context_for_testing(), false, &config, false);
+
+    EXPECT_EQ(GetParam().codec, config.codec());
+    EXPECT_EQ(GetParam().samples_per_second, config.samples_per_second());
+    EXPECT_EQ(GetParam().channel_layout, config.channel_layout());
+
+    InitializeDecoder(config);
+  }
+
+  void InitializeDecoder(const AudioDecoderConfig& config) {
+    decoder_->Initialize(config, NewExpectedStatusCB(PIPELINE_OK));

[wolenetz, 2014/06/16 23:36:08]

ToT has recent change around decoder init + decode output CB that appears to not
be included here. See 49cea86805ad5d7cf5163a480f864b61d079889e

[DaleCurtis, 2014/06/19 01:05:52]

Merged.

+    base::RunLoop().RunUntilIdle();
+  }
+
+  void Decode() {
+    ASSERT_FALSE(pending_decode_);
+    pending_decode_ = true;
+
+    AVPacket packet;
+    ASSERT_TRUE(reader_->ReadPacketForTesting(&packet));
+
+    scoped_refptr<DecoderBuffer> buffer =
+        DecoderBuffer::CopyFrom(packet.data, packet.size);
+    buffer->set_timestamp(ConvertFromTimeBase(
+        reader_->codec_context_for_testing()->time_base, packet.pts));
+    buffer->set_duration(ConvertFromTimeBase(
+        reader_->codec_context_for_testing()->time_base, packet.duration));
+
+    decoder_->Decode(
+        buffer,
+        base::Bind(&AudioDecoderTest::DecodeFinished, base::Unretained(this)));
+    base::RunLoop().RunUntilIdle();
+    av_free_packet(&packet);
+  }
+
+  void Reset() {
+    ASSERT_FALSE(pending_reset_);
+    pending_reset_ = true;
+    decoder_->Reset(
+        base::Bind(&AudioDecoderTest::ResetFinished, base::Unretained(this)));
+    base::RunLoop().RunUntilIdle();
+    ASSERT_FALSE(pending_reset_);
+  }
+
+  void Stop() {
+    decoder_->Stop();
+  }
+
+  void Seek(base::TimeDelta seek_time) {
+    Reset();
+    decoded_audio_.clear();
+    ASSERT_TRUE(reader_->SeekForTesting(seek_time));
+  }
+
+  void DecodeFinished(AudioDecoder::Status status,
+                      const scoped_refptr<AudioBuffer>& buffer) {
+    EXPECT_TRUE(pending_decode_);
+    pending_decode_ = false;
+
+    if (status == AudioDecoder::kNotEnoughData) {
+      EXPECT_FALSE(buffer);
+      Decode();
+      return;
+    }
+
+    decoded_audio_.push_back(buffer);
+
+    // If we hit a NULL buffer or have a pending reset, we expect an abort.
+    if (!buffer || pending_reset_) {
+      EXPECT_FALSE(buffer);
+      EXPECT_EQ(AudioDecoder::kAborted, status);
+      return;
+    }
+
+    EXPECT_EQ(AudioDecoder::kOk, status);
+  }
+
+  void ResetFinished() {
+    EXPECT_TRUE(pending_reset_);
+    // Reset should always finish after Decode.
+    EXPECT_FALSE(pending_decode_);
+
+    pending_reset_ = false;
+  }
+
+  // Generates an MD5 hash of the audio signal.  Should not be used for checks
+  // across platforms as audio varies slightly across platforms.
+  std::string GetDecodedAudioMD5(size_t i) {
+    CHECK_LT(i, decoded_audio_.size());
+    const scoped_refptr<AudioBuffer>& buffer = decoded_audio_[i];
+
+    scoped_ptr<AudioBus> output =
+        AudioBus::Create(buffer->channel_count(), buffer->frame_count());
+    buffer->ReadFrames(buffer->frame_count(), 0, 0, output.get());
+
+    base::MD5Context context;
+    base::MD5Init(&context);
+    for (int ch = 0; ch < output->channels(); ++ch) {
+      base::MD5Update(
+          &context,
+          base::StringPiece(reinterpret_cast<char*>(output->channel(ch)),
+                            output->frames() * sizeof(*output->channel(ch))));
+    }
+    base::MD5Digest digest;
+    base::MD5Final(&digest, &context);
+    return base::MD5DigestToBase16(digest);
+  }
+
+  void ExpectDecodedAudio(size_t i, const std::string& exact_hash) {
+    CHECK_LT(i, decoded_audio_.size());
+    const scoped_refptr<AudioBuffer>& buffer = decoded_audio_[i];
+
+    const DecodedBufferExpectations& sample_info = GetParam().expectations[i];
+    EXPECT_EQ(sample_info.timestamp, buffer->timestamp().InMicroseconds());
+    EXPECT_EQ(sample_info.duration, buffer->duration().InMicroseconds());
+    EXPECT_FALSE(buffer->end_of_stream());
+
+    scoped_ptr<AudioBus> output =
+        AudioBus::Create(buffer->channel_count(), buffer->frame_count());
+    buffer->ReadFrames(buffer->frame_count(), 0, 0, output.get());
+
+    // Generate a lossy hash of the audio used for comparison across platforms.
+    AudioHash audio_hash;
+    audio_hash.Update(output.get(), output->frames());
+    EXPECT_EQ(sample_info.hash, audio_hash.ToString());
+
+    if (!exact_hash.empty()) {
+      EXPECT_EQ(exact_hash, GetDecodedAudioMD5(i));
+
+      // Verify different hashes are being generated.  None of our test data
+      // files have audio that hashes out exactly the same.
+      if (i > 0)
+        EXPECT_NE(exact_hash, GetDecodedAudioMD5(i - 1));
+    }
+  }
+
+  void ExpectEndOfStream(size_t i) {
+    CHECK_LT(i, decoded_audio_.size());
+    EXPECT_TRUE(decoded_audio_[i]->end_of_stream());
+  }
+
+  size_t decoded_audio_size() const { return decoded_audio_.size(); }
+  base::TimeDelta start_timestamp() const { return start_timestamp_; }
+  const scoped_refptr<AudioBuffer>& decoded_audio(size_t i) {
+    return decoded_audio_[i];

[wolenetz, 2014/06/16 23:36:08]

nit: bounds-check |i|

[DaleCurtis, 2014/06/19 01:05:52]

Decline. This is a hacker style method meant to mirror vector [] access. All
callers should have checks in place for this already.

+  }
+
+ private:
+  base::MessageLoop message_loop_;
+  scoped_refptr<DecoderBuffer> data_;
+  scoped_ptr<InMemoryUrlProtocol> protocol_;
+  scoped_ptr<AudioFileReader> reader_;
+
+  scoped_ptr<AudioDecoder> decoder_;
+  bool pending_decode_;
+  bool pending_reset_;
+
+  std::deque<scoped_refptr<AudioBuffer> > decoded_audio_;
+  base::TimeDelta start_timestamp_;
+
+  DISALLOW_COPY_AND_ASSIGN(AudioDecoderTest);
+};
+
+TEST_P(AudioDecoderTest, Initialize) {
+  ASSERT_NO_FATAL_FAILURE(Initialize());
+  Stop();
+}
+
+TEST_P(AudioDecoderTest, InitializeWithNoCodecDelay) {
+  if (GetParam().decoder_type != OPUS)
+    return;
+
+  const uint8_t kOpusExtraData[] = {
+      0x4f, 0x70, 0x75, 0x73, 0x48, 0x65, 0x61, 0x64, 0x01, 0x02,
+      // The next two bytes represent the codec delay.
+      0x00, 0x00, 0x80, 0xbb, 0x00, 0x00, 0x00, 0x00, 0x00};
+  AudioDecoderConfig decoder_config;
+  decoder_config.Initialize(kCodecOpus,
+                            kSampleFormatF32,
+                            CHANNEL_LAYOUT_STEREO,
+                            48000,
+                            kOpusExtraData,
+                            ARRAYSIZE_UNSAFE(kOpusExtraData),
+                            false,
+                            false,
+                            base::TimeDelta::FromMilliseconds(80),
+                            0);
+  InitializeDecoder(decoder_config);
+  Stop();
+}
+
+// Verifies decode audio as well as the Decode() -> Reset() -> Stop() sequence.
+TEST_P(AudioDecoderTest, ProduceAudioSamples) {
+  ASSERT_NO_FATAL_FAILURE(Initialize());
+
+  // Run the test multiple times with a seek back to the beginning in between.
+  std::vector<std::string> decoded_audio_md5_hashes;
+  for (int i = 0; i < 2; ++i) {
+    for (size_t j = 0; j < kDecodeRuns; ++j) {
+      Decode();
+
+      // On the first pass record the exact MD5 hash for each decoded buffer.
+      if (i == 0)
+        decoded_audio_md5_hashes.push_back(GetDecodedAudioMD5(j));
+    }
+
+    ASSERT_EQ(kDecodeRuns, decoded_audio_size());
+
+    // On the first pass verify the basic audio hash and sample info.  On the
+    // second, verify the exact MD5 sum for each packet.  It shouldn't change.
+    for (size_t j = 0; j < kDecodeRuns; ++j) {
+      SCOPED_TRACE(base::StringPrintf("i = %d, j = %" PRIuS, i, j));
+      ExpectDecodedAudio(j, i == 0 ? "" : decoded_audio_md5_hashes[j]);
+    }
+
+    // Call one more time to trigger EOS.
+    SendEndOfStream();
+
+    ASSERT_EQ(kDecodeRuns + 1, decoded_audio_size());
+    ExpectEndOfStream(kDecodeRuns);
+
+    // Seek back to the beginning.  Calls Reset() on the decoder.
+    Seek(start_timestamp());
+  }
+
+  Stop();
+}
+
+TEST_P(AudioDecoderTest, DecodeAbort) {
+  ASSERT_NO_FATAL_FAILURE(Initialize());
+  SendAbort();
+  EXPECT_EQ(1U, decoded_audio_size());
+  EXPECT_FALSE(decoded_audio(0));
+  Stop();
+}
+
+TEST_P(AudioDecoderTest, DecodeStop) {
+  ASSERT_NO_FATAL_FAILURE(Initialize());
+  Decode();
+  EXPECT_EQ(1U, decoded_audio_size());
+  ExpectDecodedAudio(0, "");
+  Stop();
+  EXPECT_EQ(1U, decoded_audio_size());
+}
+
+const DecodedBufferExpectations kBearOpusExpectations[] = {
+    {0, 3500, "-0.26,0.87,1.36,0.84,-0.30,-1.22,"},
+    {3500, 10000, "0.09,0.23,0.21,0.03,-0.17,-0.24,"},
+    {13500, 10000, "0.10,0.24,0.23,0.04,-0.14,-0.23,"},
+};
+
+const DecoderTestData kOpusTests[] = {
+    {OPUS, kCodecOpus, "bear-opus.ogg", kBearOpusExpectations, 24, 48000,
+     CHANNEL_LAYOUT_STEREO},
+};
+
+INSTANTIATE_TEST_CASE_P(OpusAudioDecoderTest,
+                        AudioDecoderTest,
+                        testing::ValuesIn(kOpusTests));
+
+#if defined(USE_PROPRIETARY_CODECS)
+const DecodedBufferExpectations kSfxMp3Expectations[] = {
+    {0, 1065, "2.81,3.99,4.53,4.10,3.08,2.46,"},
+    {1065, 26122, "-3.81,-4.14,-3.90,-3.36,-3.03,-3.23,"},
+    {27188, 26122, "4.24,3.95,4.22,4.78,5.13,4.93,"},
+};
+
+const DecodedBufferExpectations kSfxAdtsExpectations[] = {
+    {0, 23219, "-1.90,-1.53,-0.15,1.28,1.23,-0.33,"},
+    {23219, 23219, "0.54,0.88,2.19,3.54,3.24,1.63,"},
+    {46439, 23219, "1.42,1.69,2.95,4.23,4.02,2.36,"},
+};
+#endif
+
+#if defined(OS_CHROMEOS)
+const DecodedBufferExpectations kBearFlacExpectations[] = {
+    {0, 104489, "-2.24,-0.86,0.82,2.06,1.41,-0.16,"},
+    {104489, 104489, "-1.58,-0.52,1.70,2.34,2.06,-0.05,"},
+    {208979, 104489, "-2.17,-0.75,1.03,2.14,1.41,-0.25,"},
+};
+#endif
+
+const DecodedBufferExpectations kBearOgvExpectations[] = {
+    {0, 23219, "-1.44,-1.27,0.18,1.37,1.95,0.13,"},
+    {23219, 23219, "-1.80,-1.41,-0.13,1.30,1.65,0.01,"},
+    {46439, 23219, "-1.43,-1.25,0.11,1.29,1.86,0.14,"},

[wolenetz, 2014/06/16 23:36:08]

Did bear.ogv change, are these changes vs PS2 due to ffmpeg roll, or something
else?

[DaleCurtis, 2014/06/19 01:05:52]

FFmpeg roll.

+};
+
+const DecodedBufferExpectations kSfxWaveExpectations[] = {
+    {0, 23219, "-1.23,-0.87,0.47,1.85,1.88,0.29,"},
+    {23219, 23219, "0.75,1.10,2.43,3.78,3.53,1.93,"},
+    {46439, 23219, "1.27,1.56,2.83,4.13,3.87,2.23,"},
+};
+
+const DecoderTestData kFFmpegTests[] = {
+#if defined(USE_PROPRIETARY_CODECS)
+    {FFMPEG, kCodecMP3, "sfx.mp3", kSfxMp3Expectations, 0, 44100,
+     CHANNEL_LAYOUT_MONO},
+    {FFMPEG, kCodecAAC, "sfx.adts", kSfxAdtsExpectations, 0, 44100,
+     CHANNEL_LAYOUT_MONO},
+#endif
+#if defined(OS_CHROMEOS)
+    {FFMPEG, kCodecFLAC, "bear.flac", kBearFlacExpectations, 0, 44100,
+     CHANNEL_LAYOUT_STEREO},
+#endif
+    {FFMPEG, kCodecPCM, "sfx_f32le.wav", kSfxWaveExpectations, 0, 44100,
+     CHANNEL_LAYOUT_MONO},
+    {FFMPEG, kCodecVorbis, "bear.ogv", kBearOgvExpectations, -704, 44100,

[wolenetz, 2014/06/16 23:36:08]

Negative start timestamp?! Is this normal?

[DaleCurtis, 2014/06/19 01:05:52]

Yup, I've added a comment with some more info.

+     CHANNEL_LAYOUT_STEREO},
+};
+
+INSTANTIATE_TEST_CASE_P(FFmpegAudioDecoderTest,
+                        AudioDecoderTest,
+                        testing::ValuesIn(kFFmpegTests));
+
+}  // namespace media