Merge "Merge upstream SHA 04cb763"android-cts_7.1_r1android-cts-7.1_r9android-cts-7.1_r8android-cts-7.1_r7android-cts-7.1_r6android-cts-7.1_r5android-cts-7.1_r4android-cts-7.1_r3android-cts-7.1_r29android-cts-7.1_r28android-cts-7.1_r27android-cts-7.1_r26android-cts-7.1_r25android-cts-7.1_r24android-cts-7.1_r23android-cts-7.1_r22android-cts-7.1_r21android-cts-7.1_r20android-cts-7.1_r2android-cts-7.1_r19android-cts-7.1_r18android-cts-7.1_r17android-cts-7.1_r16android-cts-7.1_r15android-cts-7.1_r14android-cts-7.1_r13android-cts-7.1_r12android-cts-7.1_r11android-cts-7.1_r10android-cts-7.1_r1android-cts-7.0_r9android-cts-7.0_r8android-cts-7.0_r7android-cts-7.0_r6android-cts-7.0_r5android-cts-7.0_r4android-cts-7.0_r33android-cts-7.0_r32android-cts-7.0_r31android-cts-7.0_r30android-cts-7.0_r3android-cts-7.0_r29android-cts-7.0_r28android-cts-7.0_r27android-cts-7.0_r26android-cts-7.0_r25android-cts-7.0_r24android-cts-7.0_r23android-cts-7.0_r22android-cts-7.0_r21android-cts-7.0_r20android-cts-7.0_r2android-cts-7.0_r19android-cts-7.0_r18android-cts-7.0_r17android-cts-7.0_r16android-cts-7.0_r15android-cts-7.0_r14android-cts-7.0_r13android-cts-7.0_r12android-cts-7.0_r11android-cts-7.0_r10android-cts-7.0_r1android-7.1.2_r9android-7.1.2_r8android-7.1.2_r6android-7.1.2_r5android-7.1.2_r4android-7.1.2_r39android-7.1.2_r38android-7.1.2_r37android-7.1.2_r36android-7.1.2_r33android-7.1.2_r32android-7.1.2_r30android-7.1.2_r3android-7.1.2_r29android-7.1.2_r28android-7.1.2_r27android-7.1.2_r25android-7.1.2_r24android-7.1.2_r23android-7.1.2_r2android-7.1.2_r19android-7.1.2_r18android-7.1.2_r17android-7.1.2_r16android-7.1.2_r15android-7.1.2_r14android-7.1.2_r13android-7.1.2_r12android-7.1.2_r11android-7.1.2_r10android-7.1.2_r1android-7.1.1_r9android-7.1.1_r8android-7.1.1_r7android-7.1.1_r61android-7.1.1_r60android-7.1.1_r6android-7.1.1_r59android-7.1.1_r58android-7.1.1_r57android-7.1.1_r56android-7.1.1_r55android-7.1.1_r54android-7.1.1_r53android-7.1.1_r52android-7.1.1_r51android-7.1.1_r50android-7.1.1_r49android-7.1.1_r48android-7.1.1_r47android-7.1.1_r46android-7.1.1_r45android-7.1.1_r44android-7.1.1_r43android-7.1.1_r42android-7.1.1_r41android-7.1.1_r40android-7.1.1_r4android-7.1.1_r39android-7.1.1_r38android-7.1.1_r35android-7.1.1_r33android-7.1.1_r32android-7.1.1_r31android-7.1.1_r3android-7.1.1_r28android-7.1.1_r27android-7.1.1_r26android-7.1.1_r25android-7.1.1_r24android-7.1.1_r23android-7.1.1_r22android-7.1.1_r21android-7.1.1_r20android-7.1.1_r2android-7.1.1_r17android-7.1.1_r16android-7.1.1_r15android-7.1.1_r14android-7.1.1_r13android-7.1.1_r12android-7.1.1_r11android-7.1.1_r10android-7.1.1_r1android-7.1.0_r7android-7.1.0_r6android-7.1.0_r5android-7.1.0_r4android-7.1.0_r3android-7.1.0_r2android-7.1.0_r1android-7.0.0_r9android-7.0.0_r8android-7.0.0_r7android-7.0.0_r6android-7.0.0_r5android-7.0.0_r4android-7.0.0_r36android-7.0.0_r35android-7.0.0_r34android-7.0.0_r33android-7.0.0_r32android-7.0.0_r31android-7.0.0_r30android-7.0.0_r3android-7.0.0_r29android-7.0.0_r28android-7.0.0_r27android-7.0.0_r24android-7.0.0_r21android-7.0.0_r19android-7.0.0_r17android-7.0.0_r15android-7.0.0_r14android-7.0.0_r13android-7.0.0_r12android-7.0.0_r11android-7.0.0_r10android-7.0.0_r1nougat-releasenougat-mr2.3-releasenougat-mr2.2-releasenougat-mr2.1-releasenougat-mr2-security-releasenougat-mr2-releasenougat-mr2-pixel-releasenougat-mr2-devnougat-mr1.8-releasenougat-mr1.7-releasenougat-mr1.6-releasenougat-mr1.5-releasenougat-mr1.4-releasenougat-mr1.3-releasenougat-mr1.2-releasenougat-mr1.1-releasenougat-mr1-volantis-releasenougat-mr1-security-releasenougat-mr1-releasenougat-mr1-flounder-releasenougat-mr1-devnougat-mr1-cts-releasenougat-mr0.5-releasenougat-dr1-releasenougat-devnougat-cts-releasenougat-bugfix-release

am: 9a337512d9

* commit '9a337512d97e37afc142dee4fd50a41b741a87d2': (797 commits)
  Add tests for verifying transport feedback for audio and video.
  Eliminate defines in talk/
  Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
  Remove assert which was incorrectly added to TcpPort::OnSentPacket.
  Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Update with new default boringssl no-aes cipher suites. Re-enable tests.
  Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )
  Re-land: "Use an explicit identifier in Config"
  Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
  Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ )
  Remove libfuzzer trybot from default trybot set.
  Add ramp-up tests for transport sequence number with and w/o audio.
  Delete remnants of non-square pixel support from cricket::VideoFrame.
  Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop().
  Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
  Re-enable tests that failed under Linux_Msan.
  Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
  Roll chromium_revision 346fea9..099be58 (369082:369139)
  Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan
  Add build_protobuf variable.
  ...

1 files changed, 284 insertions, 0 deletions

diff --git a/webrtc/modules/media_file/media_file_utility.h b/webrtc/modules/media_file/media_file_utility.h
new file mode 100644
index 0000000000..bc2fa5a2f0
--- /dev/null
+++ b/webrtc/modules/media_file/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_MEDIA_FILE_UTILITY_H_
+#define WEBRTC_MODULES_MEDIA_FILE_MEDIA_FILE_UTILITY_H_
+
+#include <stdio.h>
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/media_file/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.
+    static const size_t WAV_MAX_BUFFER_SIZE = 480 * 2 * 2;
+
+
+    int32_t InitWavCodec(uint32_t samplesPerSec,
+                         size_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,
+                           uint32_t freqInHz,
+                           size_t bytesPerSample,
+                           size_t channels,
+                           uint32_t format,
+                           size_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,
+                        size_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;
+    size_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.
+    size_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.
+    size_t _bytesPerSample;
+    size_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_MEDIA_FILE_UTILITY_H_