| Index: webkit/renderer/media/android/audio_decoder_android.cc
|
| diff --git a/webkit/renderer/media/android/audio_decoder_android.cc b/webkit/renderer/media/android/audio_decoder_android.cc
|
| deleted file mode 100644
|
| index 2d0cb1751afd6c2b9393a9889d46b835199cdbac..0000000000000000000000000000000000000000
|
| --- a/webkit/renderer/media/android/audio_decoder_android.cc
|
| +++ /dev/null
|
| @@ -1,259 +0,0 @@
|
| -// Copyright 2013 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/renderer/media/audio_decoder.h"
|
| -
|
| -#include <errno.h>
|
| -#include <fcntl.h>
|
| -#include <limits.h>
|
| -#include <sys/mman.h>
|
| -#include <unistd.h>
|
| -#include <vector>
|
| -
|
| -#include "base/callback.h"
|
| -#include "base/file_descriptor_posix.h"
|
| -#include "base/logging.h"
|
| -#include "base/posix/eintr_wrapper.h"
|
| -#include "base/shared_memory.h"
|
| -#include "media/base/android/webaudio_media_codec_info.h"
|
| -#include "media/base/audio_bus.h"
|
| -#include "media/base/limits.h"
|
| -#include "third_party/WebKit/public/platform/WebAudioBus.h"
|
| -
|
| -namespace webkit_media {
|
| -
|
| -class AudioDecoderIO {
|
| - public:
|
| - AudioDecoderIO(const char* data, size_t data_size);
|
| - ~AudioDecoderIO();
|
| - bool ShareEncodedToProcess(base::SharedMemoryHandle* handle);
|
| -
|
| - // Returns true if AudioDecoderIO was successfully created.
|
| - bool IsValid() const;
|
| -
|
| - int read_fd() const { return read_fd_; }
|
| - int write_fd() const { return write_fd_; }
|
| -
|
| - private:
|
| - // Shared memory that will hold the encoded audio data. This is
|
| - // used by MediaCodec for decoding.
|
| - base::SharedMemory encoded_shared_memory_;
|
| -
|
| - // A pipe used to communicate with MediaCodec. MediaCodec owns
|
| - // write_fd_ and writes to it.
|
| - int read_fd_;
|
| - int write_fd_;
|
| -
|
| - DISALLOW_COPY_AND_ASSIGN(AudioDecoderIO);
|
| -};
|
| -
|
| -AudioDecoderIO::AudioDecoderIO(const char* data, size_t data_size)
|
| - : read_fd_(-1),
|
| - write_fd_(-1) {
|
| -
|
| - if (!data || !data_size || data_size > 0x80000000)
|
| - return;
|
| -
|
| - // Create the shared memory and copy our data to it so that
|
| - // MediaCodec can access it.
|
| - encoded_shared_memory_.CreateAndMapAnonymous(data_size);
|
| -
|
| - if (!encoded_shared_memory_.memory())
|
| - return;
|
| -
|
| - memcpy(encoded_shared_memory_.memory(), data, data_size);
|
| -
|
| - // Create a pipe for reading/writing the decoded PCM data
|
| - int pipefd[2];
|
| -
|
| - if (pipe(pipefd))
|
| - return;
|
| -
|
| - read_fd_ = pipefd[0];
|
| - write_fd_ = pipefd[1];
|
| -}
|
| -
|
| -AudioDecoderIO::~AudioDecoderIO() {
|
| - // Close the read end of the pipe. The write end should have been
|
| - // closed by MediaCodec.
|
| - if (read_fd_ >= 0 && close(read_fd_)) {
|
| - DVLOG(1) << "Cannot close read fd " << read_fd_
|
| - << ": " << strerror(errno);
|
| - }
|
| -}
|
| -
|
| -bool AudioDecoderIO::IsValid() const {
|
| - return read_fd_ >= 0 && write_fd_ >= 0 &&
|
| - encoded_shared_memory_.memory();
|
| -}
|
| -
|
| -bool AudioDecoderIO::ShareEncodedToProcess(base::SharedMemoryHandle* handle) {
|
| - return encoded_shared_memory_.ShareToProcess(
|
| - base::Process::Current().handle(),
|
| - handle);
|
| -}
|
| -
|
| -static float ConvertSampleToFloat(int16_t sample) {
|
| - const float kMaxScale = 1.0f / std::numeric_limits<int16_t>::max();
|
| - const float kMinScale = -1.0f / std::numeric_limits<int16_t>::min();
|
| -
|
| - return sample * (sample < 0 ? kMinScale : kMaxScale);
|
| -}
|
| -
|
| -// The number of frames is known so preallocate the destination
|
| -// bus and copy the pcm data to the destination bus as it's being
|
| -// received.
|
| -static void CopyPcmDataToBus(int input_fd,
|
| - WebKit::WebAudioBus* destination_bus,
|
| - size_t number_of_frames,
|
| - unsigned number_of_channels,
|
| - double file_sample_rate) {
|
| - destination_bus->initialize(number_of_channels,
|
| - number_of_frames,
|
| - file_sample_rate);
|
| -
|
| - int16_t pipe_data[PIPE_BUF / sizeof(int16_t)];
|
| - size_t decoded_frames = 0;
|
| - ssize_t nread;
|
| -
|
| - while ((nread = HANDLE_EINTR(read(input_fd, pipe_data, sizeof(pipe_data)))) >
|
| - 0) {
|
| - size_t samples_in_pipe = nread / sizeof(int16_t);
|
| - for (size_t m = 0; m < samples_in_pipe; m += number_of_channels) {
|
| - if (decoded_frames >= number_of_frames)
|
| - break;
|
| -
|
| - for (size_t k = 0; k < number_of_channels; ++k) {
|
| - int16_t sample = pipe_data[m + k];
|
| - destination_bus->channelData(k)[decoded_frames] =
|
| - ConvertSampleToFloat(sample);
|
| - }
|
| - ++decoded_frames;
|
| - }
|
| - }
|
| -}
|
| -
|
| -// The number of frames is unknown, so keep reading and buffering
|
| -// until there's no more data and then copy the data to the
|
| -// destination bus.
|
| -static void BufferAndCopyPcmDataToBus(int input_fd,
|
| - WebKit::WebAudioBus* destination_bus,
|
| - unsigned number_of_channels,
|
| - double file_sample_rate) {
|
| - int16_t pipe_data[PIPE_BUF / sizeof(int16_t)];
|
| - std::vector<int16_t> decoded_samples;
|
| - ssize_t nread;
|
| -
|
| - while ((nread = HANDLE_EINTR(read(input_fd, pipe_data, sizeof(pipe_data)))) >
|
| - 0) {
|
| - size_t samples_in_pipe = nread / sizeof(int16_t);
|
| - if (decoded_samples.size() + samples_in_pipe > decoded_samples.capacity()) {
|
| - decoded_samples.reserve(std::max(samples_in_pipe,
|
| - 2 * decoded_samples.capacity()));
|
| - }
|
| - std::copy(pipe_data,
|
| - pipe_data + samples_in_pipe,
|
| - back_inserter(decoded_samples));
|
| - }
|
| -
|
| - DVLOG(1) << "Total samples read = " << decoded_samples.size();
|
| -
|
| - // Convert the samples and save them in the audio bus.
|
| - size_t number_of_samples = decoded_samples.size();
|
| - size_t number_of_frames = decoded_samples.size() / number_of_channels;
|
| - size_t decoded_frames = 0;
|
| -
|
| - destination_bus->initialize(number_of_channels,
|
| - number_of_frames,
|
| - file_sample_rate);
|
| -
|
| - for (size_t m = 0; m < number_of_samples; m += number_of_channels) {
|
| - for (size_t k = 0; k < number_of_channels; ++k) {
|
| - int16_t sample = decoded_samples[m + k];
|
| - destination_bus->channelData(k)[decoded_frames] =
|
| - ConvertSampleToFloat(sample);
|
| - }
|
| - ++decoded_frames;
|
| - }
|
| -}
|
| -
|
| -// To decode audio data, we want to use the Android MediaCodec class.
|
| -// But this can't run in a sandboxed process so we need initiate the
|
| -// request to MediaCodec in the browser. To do this, we create a
|
| -// shared memory buffer that holds the audio data. We send a message
|
| -// to the browser to start the decoder using this buffer and one end
|
| -// of a pipe. The MediaCodec class will decode the data from the
|
| -// shared memory and write the PCM samples back to us over a pipe.
|
| -bool DecodeAudioFileData(WebKit::WebAudioBus* destination_bus, const char* data,
|
| - size_t data_size, double sample_rate,
|
| - const WebAudioMediaCodecRunner& runner) {
|
| - AudioDecoderIO audio_decoder(data, data_size);
|
| -
|
| - if (!audio_decoder.IsValid())
|
| - return false;
|
| -
|
| - base::SharedMemoryHandle encoded_data_handle;
|
| - audio_decoder.ShareEncodedToProcess(&encoded_data_handle);
|
| - base::FileDescriptor fd(audio_decoder.write_fd(), true);
|
| -
|
| - DVLOG(1) << "DecodeAudioFileData: Starting MediaCodec";
|
| -
|
| - // Start MediaCodec processing in the browser which will read from
|
| - // encoded_data_handle for our shared memory and write the decoded
|
| - // PCM samples (16-bit integer) to our pipe.
|
| -
|
| - runner.Run(encoded_data_handle, fd, data_size);
|
| -
|
| - // First, read the number of channels, the sample rate, and the
|
| - // number of frames and a flag indicating if the file is an
|
| - // ogg/vorbis file. This must be coordinated with
|
| - // WebAudioMediaCodecBridge!
|
| - //
|
| - // TODO(rtoy): If we know the number of samples, we can create the
|
| - // destination bus directly and do the conversion directly to the
|
| - // bus instead of buffering up everything before saving the data to
|
| - // the bus.
|
| -
|
| - int input_fd = audio_decoder.read_fd();
|
| - struct media::WebAudioMediaCodecInfo info;
|
| -
|
| - DVLOG(1) << "Reading audio file info from fd " << input_fd;
|
| - ssize_t nread = HANDLE_EINTR(read(input_fd, &info, sizeof(info)));
|
| - DVLOG(1) << "read: " << nread << " bytes:\n"
|
| - << " 0: number of channels = " << info.channel_count << "\n"
|
| - << " 1: sample rate = " << info.sample_rate << "\n"
|
| - << " 2: number of frames = " << info.number_of_frames << "\n";
|
| -
|
| - if (nread != sizeof(info))
|
| - return false;
|
| -
|
| - unsigned number_of_channels = info.channel_count;
|
| - double file_sample_rate = static_cast<double>(info.sample_rate);
|
| - size_t number_of_frames = info.number_of_frames;
|
| -
|
| - // Sanity checks
|
| - if (!number_of_channels ||
|
| - number_of_channels > media::limits::kMaxChannels ||
|
| - file_sample_rate < media::limits::kMinSampleRate ||
|
| - file_sample_rate > media::limits::kMaxSampleRate) {
|
| - return false;
|
| - }
|
| -
|
| - if (number_of_frames > 0) {
|
| - CopyPcmDataToBus(input_fd,
|
| - destination_bus,
|
| - number_of_frames,
|
| - number_of_channels,
|
| - file_sample_rate);
|
| - } else {
|
| - BufferAndCopyPcmDataToBus(input_fd,
|
| - destination_bus,
|
| - number_of_channels,
|
| - file_sample_rate);
|
| - }
|
| -
|
| - return true;
|
| -}
|
| -
|
| -} // namespace webkit_media
|
|
|