Merge changes I7bbf776e,I1b827825

am: fe8b4a6579

* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
  WIP: Changes after merge commit 'cb3f9bd'
  Make the nonlinear beamformer steerable
  Utilize bitrate above codec max to protect video.
  Enable VP9 internal resize by default.
  Filter overlapping RTP header extensions.
  Make VCMEncodedFrameCallback const.
  MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
  Remove redudant encoder rate calls.
  Create isolate files for nonparallel tests.
  Register header extensions in RtpRtcpObserver to avoid log spam.
  Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
  ACM: Move NACK functionality inside NetEq
  Fix chromium-style warnings in webrtc/sound/.
  Create a 'webrtc_nonparallel_tests' target.
  Update scalability structure data according to updates in the RTP payload profile.
  audio_coding: rename interface -> include
  Rewrote perform_action_on_all_files to be parallell.
  Update reference indices according to updates in the RTP payload profile.
  Disable P2PTransport...TestFailoverControlledSide on Memcheck
  pass clangcl compile options to ignore warnings in gflags.cc
  ...

1 files changed, 284 insertions, 0 deletions

diff --git a/webrtc/modules/media_file/source/media_file_utility.h b/webrtc/modules/media_file/source/media_file_utility.h
new file mode 100644
index 0000000000..2823ceca8a
--- /dev/null
+++ b/webrtc/modules/media_file/source/media_file_utility.h
@@ -0,0 +1,284 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Note: the class cannot be used for reading and writing at the same time.
+#ifndef WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_
+#define WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_
+
+#include <stdio.h>
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/media_file/interface/media_file_defines.h"
+
+namespace webrtc {
+class InStream;
+class OutStream;
+
+class ModuleFileUtility
+{
+public:
+
+    ModuleFileUtility(const int32_t id);
+    ~ModuleFileUtility();
+
+    // Prepare for playing audio from stream.
+    // startPointMs and stopPointMs, unless zero, specify what part of the file
+    // should be read. From startPointMs ms to stopPointMs ms.
+    int32_t InitWavReading(InStream& stream,
+                           const uint32_t startPointMs = 0,
+                           const uint32_t stopPointMs = 0);
+
+    // Put 10-60ms of audio data from stream into the audioBuffer depending on
+    // codec frame size. dataLengthInBytes indicates the size of audioBuffer.
+    // The return value is the number of bytes written to audioBuffer.
+    // Note: This API only play mono audio but can be used on file containing
+    // audio with more channels (in which case the audio will be converted to
+    // mono).
+    int32_t ReadWavDataAsMono(InStream& stream, int8_t* audioBuffer,
+                              const size_t dataLengthInBytes);
+
+    // Put 10-60ms, depending on codec frame size, of audio data from file into
+    // audioBufferLeft and audioBufferRight. The buffers contain the left and
+    // right channel of played out stereo audio.
+    // dataLengthInBytes  indicates the size of both audioBufferLeft and
+    // audioBufferRight.
+    // The return value is the number of bytes read for each buffer.
+    // Note: This API can only be successfully called for WAV files with stereo
+    // audio.
+    int32_t ReadWavDataAsStereo(InStream& wav,
+                                int8_t* audioBufferLeft,
+                                int8_t* audioBufferRight,
+                                const size_t bufferLength);
+
+    // Prepare for recording audio to stream.
+    // codecInst specifies the encoding of the audio data.
+    // Note: codecInst.channels should be set to 2 for stereo (and 1 for
+    // mono). Stereo is only supported for WAV files.
+    int32_t InitWavWriting(OutStream& stream, const CodecInst& codecInst);
+
+    // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
+    // to file. The audio frame size is determined by the codecInst.pacsize
+    // parameter of the last sucessfull StartRecordingAudioFile(..) call.
+    // The return value is the number of bytes written to audioBuffer.
+    int32_t WriteWavData(OutStream& stream,
+                         const int8_t* audioBuffer,
+                         const size_t bufferLength);
+
+    // Finalizes the WAV header so that it is correct if nothing more will be
+    // written to stream.
+    // Note: this API must be called before closing stream to ensure that the
+    //       WAVE header is updated with the file size. Don't call this API
+    //       if more samples are to be written to stream.
+    int32_t UpdateWavHeader(OutStream& stream);
+
+    // Prepare for playing audio from stream.
+    // startPointMs and stopPointMs, unless zero, specify what part of the file
+    // should be read. From startPointMs ms to stopPointMs ms.
+    // freqInHz is the PCM sampling frequency.
+    // NOTE, allowed frequencies are 8000, 16000 and 32000 (Hz)
+    int32_t InitPCMReading(InStream& stream,
+                           const uint32_t startPointMs = 0,
+                           const uint32_t stopPointMs = 0,
+                           const uint32_t freqInHz = 16000);
+
+    // Put 10-60ms of audio data from stream into the audioBuffer depending on
+    // codec frame size. dataLengthInBytes indicates the size of audioBuffer.
+    // The return value is the number of bytes written to audioBuffer.
+    int32_t ReadPCMData(InStream& stream, int8_t* audioBuffer,
+                        const size_t dataLengthInBytes);
+
+    // Prepare for recording audio to stream.
+    // freqInHz is the PCM sampling frequency.
+    // NOTE, allowed frequencies are 8000, 16000 and 32000 (Hz)
+    int32_t InitPCMWriting(OutStream& stream, const uint32_t freqInHz = 16000);
+
+    // Write one 10ms audio frame, i.e. the bufferLength first bytes of
+    // audioBuffer, to file. The audio frame size is determined by the freqInHz
+    // parameter of the last sucessfull InitPCMWriting(..) call.
+    // The return value is the number of bytes written to audioBuffer.
+    int32_t WritePCMData(OutStream& stream,
+                         const int8_t* audioBuffer,
+                         size_t bufferLength);
+
+    // Prepare for playing audio from stream.
+    // startPointMs and stopPointMs, unless zero, specify what part of the file
+    // should be read. From startPointMs ms to stopPointMs ms.
+    int32_t InitCompressedReading(InStream& stream,
+                                  const uint32_t startPointMs = 0,
+                                  const uint32_t stopPointMs = 0);
+
+    // Put 10-60ms of audio data from stream into the audioBuffer depending on
+    // codec frame size. dataLengthInBytes indicates the size of audioBuffer.
+    // The return value is the number of bytes written to audioBuffer.
+    int32_t ReadCompressedData(InStream& stream,
+                               int8_t* audioBuffer,
+                               const size_t dataLengthInBytes);
+
+    // Prepare for recording audio to stream.
+    // codecInst specifies the encoding of the audio data.
+    int32_t InitCompressedWriting(OutStream& stream,
+                                  const CodecInst& codecInst);
+
+    // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
+    // to file. The audio frame size is determined by the codecInst.pacsize
+    // parameter of the last sucessfull InitCompressedWriting(..) call.
+    // The return value is the number of bytes written to stream.
+    // Note: bufferLength must be exactly one frame.
+    int32_t WriteCompressedData(OutStream& stream,
+                                const int8_t* audioBuffer,
+                                const size_t bufferLength);
+
+    // Prepare for playing audio from stream.
+    // codecInst specifies the encoding of the audio data.
+    int32_t InitPreEncodedReading(InStream& stream,
+                                  const CodecInst& codecInst);
+
+    // Put 10-60ms of audio data from stream into the audioBuffer depending on
+    // codec frame size. dataLengthInBytes indicates the size of audioBuffer.
+    // The return value is the number of bytes written to audioBuffer.
+    int32_t ReadPreEncodedData(InStream& stream,
+                               int8_t* audioBuffer,
+                               const size_t dataLengthInBytes);
+
+    // Prepare for recording audio to stream.
+    // codecInst specifies the encoding of the audio data.
+    int32_t InitPreEncodedWriting(OutStream& stream,
+                                  const CodecInst& codecInst);
+
+    // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
+    // to stream. The audio frame size is determined by the codecInst.pacsize
+    // parameter of the last sucessfull InitPreEncodedWriting(..) call.
+   // The return value is the number of bytes written to stream.
+    // Note: bufferLength must be exactly one frame.
+    int32_t WritePreEncodedData(OutStream& stream,
+                                const int8_t* inData,
+                                const size_t dataLengthInBytes);
+
+    // Set durationMs to the size of the file (in ms) specified by fileName.
+    // freqInHz specifies the sampling frequency of the file.
+    int32_t FileDurationMs(const char* fileName,
+                           const FileFormats fileFormat,
+                           const uint32_t freqInHz = 16000);
+
+    // Return the number of ms that have been played so far.
+    uint32_t PlayoutPositionMs();
+
+    // Update codecInst according to the current audio codec being used for
+    // reading or writing.
+    int32_t codec_info(CodecInst& codecInst);
+
+private:
+    // Biggest WAV frame supported is 10 ms at 48kHz of 2 channel, 16 bit audio.
+    enum{WAV_MAX_BUFFER_SIZE = 480*2*2};
+
+
+    int32_t InitWavCodec(uint32_t samplesPerSec,
+                         uint32_t channels,
+                         uint32_t bitsPerSample,
+                         uint32_t formatTag);
+
+    // Parse the WAV header in stream.
+    int32_t ReadWavHeader(InStream& stream);
+
+    // Update the WAV header. freqInHz, bytesPerSample, channels, format,
+    // lengthInBytes specify characterists of the audio data.
+    // freqInHz is the sampling frequency. bytesPerSample is the sample size in
+    // bytes. channels is the number of channels, e.g. 1 is mono and 2 is
+    // stereo. format is the encode format (e.g. PCMU, PCMA, PCM etc).
+    // lengthInBytes is the number of bytes the audio samples are using up.
+    int32_t WriteWavHeader(OutStream& stream,
+                           const uint32_t freqInHz,
+                           const uint32_t bytesPerSample,
+                           const uint32_t channels,
+                           const uint32_t format,
+                           const uint32_t lengthInBytes);
+
+    // Put dataLengthInBytes of audio data from stream into the audioBuffer.
+    // The return value is the number of bytes written to audioBuffer.
+    int32_t ReadWavData(InStream& stream, uint8_t* audioBuffer,
+                        const uint32_t dataLengthInBytes);
+
+    // Update the current audio codec being used for reading or writing
+    // according to codecInst.
+    int32_t set_codec_info(const CodecInst& codecInst);
+
+    struct WAVE_FMTINFO_header
+    {
+        int16_t formatTag;
+        int16_t nChannels;
+        int32_t nSamplesPerSec;
+        int32_t nAvgBytesPerSec;
+        int16_t nBlockAlign;
+        int16_t nBitsPerSample;
+    };
+    // Identifiers for preencoded files.
+    enum MediaFileUtility_CodecType
+    {
+        kCodecNoCodec  = 0,
+        kCodecIsac,
+        kCodecIsacSwb,
+        kCodecIsacLc,
+        kCodecL16_8Khz,
+        kCodecL16_16kHz,
+        kCodecL16_32Khz,
+        kCodecPcmu,
+        kCodecPcma,
+        kCodecIlbc20Ms,
+        kCodecIlbc30Ms,
+        kCodecG722,
+        kCodecG722_1_32Kbps,
+        kCodecG722_1_24Kbps,
+        kCodecG722_1_16Kbps,
+        kCodecG722_1c_48,
+        kCodecG722_1c_32,
+        kCodecG722_1c_24,
+        kCodecAmr,
+        kCodecAmrWb,
+        kCodecG729,
+        kCodecG729_1,
+        kCodecG726_40,
+        kCodecG726_32,
+        kCodecG726_24,
+        kCodecG726_16,
+        kCodecSpeex8Khz,
+        kCodecSpeex16Khz
+    };
+
+    // TODO (hellner): why store multiple formats. Just store either codec_info_
+    //                 or _wavFormatObj and supply conversion functions.
+    WAVE_FMTINFO_header _wavFormatObj;
+    int32_t _dataSize;      // Chunk size if reading a WAV file
+    // Number of bytes to read. I.e. frame size in bytes. May be multiple
+    // chunks if reading WAV.
+    int32_t _readSizeBytes;
+
+    int32_t _id;
+
+    uint32_t _stopPointInMs;
+    uint32_t _startPointInMs;
+    uint32_t _playoutPositionMs;
+    size_t _bytesWritten;
+
+    CodecInst codec_info_;
+    MediaFileUtility_CodecType _codecId;
+
+    // The amount of bytes, on average, used for one audio sample.
+    int32_t  _bytesPerSample;
+    int32_t  _readPos;
+
+    // Only reading or writing can be enabled, not both.
+    bool _reading;
+    bool _writing;
+
+    // Scratch buffer used for turning stereo audio to mono.
+    uint8_t _tempData[WAV_MAX_BUFFER_SIZE];
+};
+}  // namespace webrtc
+#endif // WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_