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| 1 // Copyright 2013 The Chromium Authors. All rights reserved. | |
| 2 // Use of this source code is governed by a BSD-style license that can be | |
| 3 // found in the LICENSE file. | |
| 4 | |
| 5 #include "webkit/renderer/media/audio_decoder.h" | |
| 6 | |
| 7 #include <errno.h> | |
| 8 #include <fcntl.h> | |
| 9 #include <limits.h> | |
| 10 #include <sys/mman.h> | |
| 11 #include <unistd.h> | |
| 12 #include <vector> | |
| 13 | |
| 14 #include "base/callback.h" | |
| 15 #include "base/file_descriptor_posix.h" | |
| 16 #include "base/logging.h" | |
| 17 #include "base/posix/eintr_wrapper.h" | |
| 18 #include "base/shared_memory.h" | |
| 19 #include "media/base/android/webaudio_media_codec_info.h" | |
| 20 #include "media/base/audio_bus.h" | |
| 21 #include "media/base/limits.h" | |
| 22 #include "third_party/WebKit/public/platform/WebAudioBus.h" | |
| 23 | |
| 24 namespace webkit_media { | |
| 25 | |
| 26 class AudioDecoderIO { | |
| 27 public: | |
| 28 AudioDecoderIO(const char* data, size_t data_size); | |
| 29 ~AudioDecoderIO(); | |
| 30 bool ShareEncodedToProcess(base::SharedMemoryHandle* handle); | |
| 31 | |
| 32 // Returns true if AudioDecoderIO was successfully created. | |
| 33 bool IsValid() const; | |
| 34 | |
| 35 int read_fd() const { return read_fd_; } | |
| 36 int write_fd() const { return write_fd_; } | |
| 37 | |
| 38 private: | |
| 39 // Shared memory that will hold the encoded audio data. This is | |
| 40 // used by MediaCodec for decoding. | |
| 41 base::SharedMemory encoded_shared_memory_; | |
| 42 | |
| 43 // A pipe used to communicate with MediaCodec. MediaCodec owns | |
| 44 // write_fd_ and writes to it. | |
| 45 int read_fd_; | |
| 46 int write_fd_; | |
| 47 | |
| 48 DISALLOW_COPY_AND_ASSIGN(AudioDecoderIO); | |
| 49 }; | |
| 50 | |
| 51 AudioDecoderIO::AudioDecoderIO(const char* data, size_t data_size) | |
| 52 : read_fd_(-1), | |
| 53 write_fd_(-1) { | |
| 54 | |
| 55 if (!data || !data_size || data_size > 0x80000000) | |
| 56 return; | |
| 57 | |
| 58 // Create the shared memory and copy our data to it so that | |
| 59 // MediaCodec can access it. | |
| 60 encoded_shared_memory_.CreateAndMapAnonymous(data_size); | |
| 61 | |
| 62 if (!encoded_shared_memory_.memory()) | |
| 63 return; | |
| 64 | |
| 65 memcpy(encoded_shared_memory_.memory(), data, data_size); | |
| 66 | |
| 67 // Create a pipe for reading/writing the decoded PCM data | |
| 68 int pipefd[2]; | |
| 69 | |
| 70 if (pipe(pipefd)) | |
| 71 return; | |
| 72 | |
| 73 read_fd_ = pipefd[0]; | |
| 74 write_fd_ = pipefd[1]; | |
| 75 } | |
| 76 | |
| 77 AudioDecoderIO::~AudioDecoderIO() { | |
| 78 // Close the read end of the pipe. The write end should have been | |
| 79 // closed by MediaCodec. | |
| 80 if (read_fd_ >= 0 && close(read_fd_)) { | |
| 81 DVLOG(1) << "Cannot close read fd " << read_fd_ | |
| 82 << ": " << strerror(errno); | |
| 83 } | |
| 84 } | |
| 85 | |
| 86 bool AudioDecoderIO::IsValid() const { | |
| 87 return read_fd_ >= 0 && write_fd_ >= 0 && | |
| 88 encoded_shared_memory_.memory(); | |
| 89 } | |
| 90 | |
| 91 bool AudioDecoderIO::ShareEncodedToProcess(base::SharedMemoryHandle* handle) { | |
| 92 return encoded_shared_memory_.ShareToProcess( | |
| 93 base::Process::Current().handle(), | |
| 94 handle); | |
| 95 } | |
| 96 | |
| 97 static float ConvertSampleToFloat(int16_t sample) { | |
| 98 const float kMaxScale = 1.0f / std::numeric_limits<int16_t>::max(); | |
| 99 const float kMinScale = -1.0f / std::numeric_limits<int16_t>::min(); | |
| 100 | |
| 101 return sample * (sample < 0 ? kMinScale : kMaxScale); | |
| 102 } | |
| 103 | |
| 104 // The number of frames is known so preallocate the destination | |
| 105 // bus and copy the pcm data to the destination bus as it's being | |
| 106 // received. | |
| 107 static void CopyPcmDataToBus(int input_fd, | |
| 108 WebKit::WebAudioBus* destination_bus, | |
| 109 size_t number_of_frames, | |
| 110 unsigned number_of_channels, | |
| 111 double file_sample_rate) { | |
| 112 destination_bus->initialize(number_of_channels, | |
| 113 number_of_frames, | |
| 114 file_sample_rate); | |
| 115 | |
| 116 int16_t pipe_data[PIPE_BUF / sizeof(int16_t)]; | |
| 117 size_t decoded_frames = 0; | |
| 118 ssize_t nread; | |
| 119 | |
| 120 while ((nread = HANDLE_EINTR(read(input_fd, pipe_data, sizeof(pipe_data)))) > | |
| 121 0) { | |
| 122 size_t samples_in_pipe = nread / sizeof(int16_t); | |
| 123 for (size_t m = 0; m < samples_in_pipe; m += number_of_channels) { | |
| 124 if (decoded_frames >= number_of_frames) | |
| 125 break; | |
| 126 | |
| 127 for (size_t k = 0; k < number_of_channels; ++k) { | |
| 128 int16_t sample = pipe_data[m + k]; | |
| 129 destination_bus->channelData(k)[decoded_frames] = | |
| 130 ConvertSampleToFloat(sample); | |
| 131 } | |
| 132 ++decoded_frames; | |
| 133 } | |
| 134 } | |
| 135 } | |
| 136 | |
| 137 // The number of frames is unknown, so keep reading and buffering | |
| 138 // until there's no more data and then copy the data to the | |
| 139 // destination bus. | |
| 140 static void BufferAndCopyPcmDataToBus(int input_fd, | |
| 141 WebKit::WebAudioBus* destination_bus, | |
| 142 unsigned number_of_channels, | |
| 143 double file_sample_rate) { | |
| 144 int16_t pipe_data[PIPE_BUF / sizeof(int16_t)]; | |
| 145 std::vector<int16_t> decoded_samples; | |
| 146 ssize_t nread; | |
| 147 | |
| 148 while ((nread = HANDLE_EINTR(read(input_fd, pipe_data, sizeof(pipe_data)))) > | |
| 149 0) { | |
| 150 size_t samples_in_pipe = nread / sizeof(int16_t); | |
| 151 if (decoded_samples.size() + samples_in_pipe > decoded_samples.capacity()) { | |
| 152 decoded_samples.reserve(std::max(samples_in_pipe, | |
| 153 2 * decoded_samples.capacity())); | |
| 154 } | |
| 155 std::copy(pipe_data, | |
| 156 pipe_data + samples_in_pipe, | |
| 157 back_inserter(decoded_samples)); | |
| 158 } | |
| 159 | |
| 160 DVLOG(1) << "Total samples read = " << decoded_samples.size(); | |
| 161 | |
| 162 // Convert the samples and save them in the audio bus. | |
| 163 size_t number_of_samples = decoded_samples.size(); | |
| 164 size_t number_of_frames = decoded_samples.size() / number_of_channels; | |
| 165 size_t decoded_frames = 0; | |
| 166 | |
| 167 destination_bus->initialize(number_of_channels, | |
| 168 number_of_frames, | |
| 169 file_sample_rate); | |
| 170 | |
| 171 for (size_t m = 0; m < number_of_samples; m += number_of_channels) { | |
| 172 for (size_t k = 0; k < number_of_channels; ++k) { | |
| 173 int16_t sample = decoded_samples[m + k]; | |
| 174 destination_bus->channelData(k)[decoded_frames] = | |
| 175 ConvertSampleToFloat(sample); | |
| 176 } | |
| 177 ++decoded_frames; | |
| 178 } | |
| 179 } | |
| 180 | |
| 181 // To decode audio data, we want to use the Android MediaCodec class. | |
| 182 // But this can't run in a sandboxed process so we need initiate the | |
| 183 // request to MediaCodec in the browser. To do this, we create a | |
| 184 // shared memory buffer that holds the audio data. We send a message | |
| 185 // to the browser to start the decoder using this buffer and one end | |
| 186 // of a pipe. The MediaCodec class will decode the data from the | |
| 187 // shared memory and write the PCM samples back to us over a pipe. | |
| 188 bool DecodeAudioFileData(WebKit::WebAudioBus* destination_bus, const char* data, | |
| 189 size_t data_size, double sample_rate, | |
| 190 const WebAudioMediaCodecRunner& runner) { | |
| 191 AudioDecoderIO audio_decoder(data, data_size); | |
| 192 | |
| 193 if (!audio_decoder.IsValid()) | |
| 194 return false; | |
| 195 | |
| 196 base::SharedMemoryHandle encoded_data_handle; | |
| 197 audio_decoder.ShareEncodedToProcess(&encoded_data_handle); | |
| 198 base::FileDescriptor fd(audio_decoder.write_fd(), true); | |
| 199 | |
| 200 DVLOG(1) << "DecodeAudioFileData: Starting MediaCodec"; | |
| 201 | |
| 202 // Start MediaCodec processing in the browser which will read from | |
| 203 // encoded_data_handle for our shared memory and write the decoded | |
| 204 // PCM samples (16-bit integer) to our pipe. | |
| 205 | |
| 206 runner.Run(encoded_data_handle, fd, data_size); | |
| 207 | |
| 208 // First, read the number of channels, the sample rate, and the | |
| 209 // number of frames and a flag indicating if the file is an | |
| 210 // ogg/vorbis file. This must be coordinated with | |
| 211 // WebAudioMediaCodecBridge! | |
| 212 // | |
| 213 // TODO(rtoy): If we know the number of samples, we can create the | |
| 214 // destination bus directly and do the conversion directly to the | |
| 215 // bus instead of buffering up everything before saving the data to | |
| 216 // the bus. | |
| 217 | |
| 218 int input_fd = audio_decoder.read_fd(); | |
| 219 struct media::WebAudioMediaCodecInfo info; | |
| 220 | |
| 221 DVLOG(1) << "Reading audio file info from fd " << input_fd; | |
| 222 ssize_t nread = HANDLE_EINTR(read(input_fd, &info, sizeof(info))); | |
| 223 DVLOG(1) << "read: " << nread << " bytes:\n" | |
| 224 << " 0: number of channels = " << info.channel_count << "\n" | |
| 225 << " 1: sample rate = " << info.sample_rate << "\n" | |
| 226 << " 2: number of frames = " << info.number_of_frames << "\n"; | |
| 227 | |
| 228 if (nread != sizeof(info)) | |
| 229 return false; | |
| 230 | |
| 231 unsigned number_of_channels = info.channel_count; | |
| 232 double file_sample_rate = static_cast<double>(info.sample_rate); | |
| 233 size_t number_of_frames = info.number_of_frames; | |
| 234 | |
| 235 // Sanity checks | |
| 236 if (!number_of_channels || | |
| 237 number_of_channels > media::limits::kMaxChannels || | |
| 238 file_sample_rate < media::limits::kMinSampleRate || | |
| 239 file_sample_rate > media::limits::kMaxSampleRate) { | |
| 240 return false; | |
| 241 } | |
| 242 | |
| 243 if (number_of_frames > 0) { | |
| 244 CopyPcmDataToBus(input_fd, | |
| 245 destination_bus, | |
| 246 number_of_frames, | |
| 247 number_of_channels, | |
| 248 file_sample_rate); | |
| 249 } else { | |
| 250 BufferAndCopyPcmDataToBus(input_fd, | |
| 251 destination_bus, | |
| 252 number_of_channels, | |
| 253 file_sample_rate); | |
| 254 } | |
| 255 | |
| 256 return true; | |
| 257 } | |
| 258 | |
| 259 } // namespace webkit_media | |
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