| /frameworks/av/media/libstagefright/ |
| AwesomePlayer.cpp | 393 // Attempt to approximate overall stream bitrate by summing all
394 // tracks' individual bitrates, if not all of them advertise bitrate,
403 int32_t bitrate; local
404 if (!meta->findInt32(kKeyBitRate, &bitrate)) {
407 ALOGV("track of type '%s' does not publish bitrate", mime);
413 totalBitRate += bitrate;
651 bool AwesomePlayer::getBitrate(int64_t *bitrate) {
655 *bitrate = size * 8000000ll / mDurationUs; // in bits/sec
660 *bitrate = mBitrate;
664 *bitrate = 0
671 int64_t bitrate; local
[all...] |
| /external/chromium_org/media/cast/audio_sender/ |
| audio_encoder_unittest.cc | 153 audio_config.bitrate = kDefaultAudioEncoderBitrate;
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| audio_sender.cc | 47 configured_encoder_bitrate_(audio_config.bitrate),
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| /external/chromium_org/media/cast/rtcp/ |
| rtcp_utility.h | 133 uint32 bitrate; member in struct:media::cast::RtcpFieldPayloadSpecificRembItem
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| /external/chromium_org/media/filters/ |
| ffmpeg_demuxer.cc | 544 // Helper for calculating the bitrate of the media based on information stored
547 // Returns 0 if a bitrate could not be determined.
552 // If there is a bitrate set on the container, use it.
557 int bitrate = 0; local
560 bitrate += codec_context->bit_rate;
562 if (bitrate > 0)
563 return bitrate;
565 // See if we can approximate the bitrate as long as we have a filesize and
723 // Good to go: set the duration and bitrate and notify we're done
789 media_log_->SetIntegerProperty("bitrate", bitrate_)
[all...] |
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
| rtp_rtcp.h | 300 * get sent bitrate in Kbit/s
308 * Called on any new send bitrate estimate.
547 * (REMB) Receiver Estimated Max Bitrate
553 virtual int32_t SetREMBData(const uint32_t bitrate,
701 * Set the target send bitrate
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| rtp_rtcp_defines.h | 285 virtual void OnReceivedEstimatedBitrate(const uint32_t bitrate) = 0;
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| rtcp_sender.h | 140 int32_t SetREMBData(const uint32_t bitrate,
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| rtp_rtcp_impl.h | 203 // (REMB) Receiver Estimated Max Bitrate.
208 virtual int32_t SetREMBData(const uint32_t bitrate,
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| rtp_sender.h | 22 #include "webrtc/modules/rtp_rtcp/source/bitrate.h"
67 class RTPSender : public RTPSenderInterface, public Bitrate::Observer {
86 void SetTargetBitrate(uint32_t bitrate);
272 // Called on new send bitrate estimate.
334 Bitrate bitrate_sent_;
360 Bitrate nack_bitrate_;
398 // that the target bitrate is still valid.
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| /external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
| video_coding_impl.h | 76 int Bitrate(unsigned int* bitrate) const;
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| jitter_buffer_unittest.cc | 1639 uint32_t bitrate = 0; local
[all...] |
| /external/chromium_org/third_party/webrtc/video_engine/test/auto_test/source/ |
| vie_autotest_codec.cc | 78 const unsigned int bitrate) {
81 last_incoming_bitrate_ += bitrate;
97 const unsigned int bitrate) {
100 last_outgoing_bitrate_ += bitrate;
[all...] |
| /frameworks/av/media/libstagefright/codecs/avc/enc/src/ |
| avcenc_int.h | 170 int bitrate; /* bitrate for current frame */ member in struct:tagMultiPass
196 /* For target bitrate or framerate update */
197 OsclFloat target_bits_per_frame; /* = C = bitrate/framerate */
219 /* bitrate-robustness tradeoff */
240 int32 bitRate; /* target bit rate for the overall clip in bits/second*/
313 int max_BitVariance_num; /* the number of the maximum bit variance within the given buffer with the unit of 10% of bitrate/framerate*/
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| avcenc_api.cpp | 684 OSCL_EXPORT_REF AVCEnc_Status PVAVCEncUpdateBitRate(AVCHandle *avcHandle, uint32 bitrate)
687 OSCL_UNUSED_ARG(bitrate);
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| /hardware/qcom/media/mm-video-v4l2/vidc/venc/inc/ |
| video_encoder_device_v4l2.h | 309 struct msm_venc_targetbitrate bitrate; member in class:venc_dev
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| /hardware/qcom/msm8960/kernel-headers/linux/ |
| msm_q6venc.h | 164 u32 bitrate; member in struct:venc_common_config
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| /hardware/qcom/msm8960/original-kernel-headers/linux/ |
| msm_q6venc.h | 133 u32 bitrate; member in struct:venc_common_config
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| /external/chromium_org/net/quic/ |
| quic_protocol.cc | 528 << congestion_frame.fix_rate.bitrate.ToBytesPerSecond();
545 : bitrate(QuicBandwidth::Zero()) {
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| /external/chromium_org/third_party/webrtc/modules/bitrate_controller/ |
| bitrate_controller_unittest.cc | 46 virtual void OnNetworkChanged(const uint32_t bitrate,
49 last_bitrate_ = bitrate;
108 // Receive a high remb, test bitrate inc.
114 // Test bitrate increase 8% per second.
173 // Receive a high remb, test bitrate inc.
179 // Test start bitrate.
189 // Test bitrate increase 8% per second.
267 // Receive a high REMB, test bitrate increase.
273 // Ramp up to max bitrate.
291 // Packet loss on the first stream. Verify that bitrate decreases
[all...] |
| /external/chromium_org/third_party/webrtc/video_engine/ |
| vie_codec_impl.cc | 169 LOG(LS_INFO) << "New max bitrate set " << video_codec_internal.maxBitrate;
397 unsigned int* bitrate) const {
404 return vie_encoder->CodecTargetBitrate(static_cast<uint32_t*>(bitrate));
652 LOG(LS_ERROR) << "Invalid start bitrate.";
656 LOG(LS_ERROR) << "Invalid min bitrate.";
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| /external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/test/ |
| bwe_test.cc | 122 unsigned int bitrate) {
291 // Allocate the bitrate evenly between the senders.
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/mocks/ |
| mock_rtp_rtcp.h | 188 int32_t(const uint32_t bitrate, const uint8_t numberOfSSRC, const uint32_t* SSRC));
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| /frameworks/av/media/libstagefright/codecs/m4v_h263/enc/src/ |
| mp4lib_int.h | 182 Int LayerBitRate[4]; /* Encoded BitRate */
183 Int LayerMaxBitRate[4]; /* Maximum Encoded BitRate */
269 Int bitrate; /* bitrate for current frame */ member in struct:tagMultiPass
295 /* For target bitrate or framerate update */
296 float target_bits_per_frame; /* = C = bitrate/framerate */
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| /frameworks/av/media/libstagefright/include/ |
| AwesomePlayer.h | 206 int64_t mBitrate; // total bitrate of the file (in bps) or -1 if unknown.
299 bool getBitrate(int64_t *bitrate);
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