Clementine-audio-player-Mac.../ext/clementine-spotifyblob/mediapipeline.cpp

169 lines
5.2 KiB
C++

/* This file is part of Clementine.
Copyright 2011, David Sansome <me@davidsansome.com>
Licensed under the Apache License, Version 2.0 (the "License");
you may not use this file except in compliance with the License.
You may obtain a copy of the License at
http://www.apache.org/licenses/LICENSE-2.0
Unless required by applicable law or agreed to in writing, software
distributed under the License is distributed on an "AS IS" BASIS,
WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
See the License for the specific language governing permissions and
limitations under the License.
*/
// Note: this file is licensed under the Apache License instead of GPL because
// it is used by the Spotify blob which links against libspotify and is not GPL
// compatible.
#include "mediapipeline.h"
#include "core/logging.h"
#include "core/timeconstants.h"
#include <cstring>
MediaPipeline::MediaPipeline(int port, quint64 length_msec)
: port_(port),
length_msec_(length_msec),
accepting_data_(true),
pipeline_(nullptr),
appsrc_(nullptr),
byte_rate_(1),
offset_bytes_(0) {}
MediaPipeline::~MediaPipeline() {
if (pipeline_) {
gst_element_set_state(pipeline_, GST_STATE_NULL);
gst_object_unref(GST_OBJECT(pipeline_));
}
}
bool MediaPipeline::Init(int sample_rate, int channels) {
if (is_initialised()) return false;
pipeline_ = gst_pipeline_new("pipeline");
// Create elements
appsrc_ = GST_APP_SRC(gst_element_factory_make("appsrc", nullptr));
GstElement* gdppay = gst_element_factory_make("gdppay", nullptr);
tcpsink_ = gst_element_factory_make("tcpclientsink", nullptr);
if (!pipeline_ || !appsrc_ || !tcpsink_) {
if (pipeline_) {
gst_object_unref(GST_OBJECT(pipeline_));
pipeline_ = nullptr;
}
if (appsrc_) {
gst_object_unref(GST_OBJECT(appsrc_));
appsrc_ = nullptr;
}
if (gdppay) {
gst_object_unref(GST_OBJECT(gdppay));
}
if (tcpsink_) {
gst_object_unref(GST_OBJECT(tcpsink_));
tcpsink_ = nullptr;
}
return false;
}
// Add elements to the pipeline and link them
gst_bin_add(GST_BIN(pipeline_), GST_ELEMENT(appsrc_));
gst_bin_add(GST_BIN(pipeline_), gdppay);
gst_bin_add(GST_BIN(pipeline_), tcpsink_);
gst_element_link_many(GST_ELEMENT(appsrc_), gdppay, tcpsink_, nullptr);
// Set the sink's port
g_object_set(G_OBJECT(tcpsink_), "host", "127.0.0.1", nullptr);
g_object_set(G_OBJECT(tcpsink_), "port", port_, nullptr);
// Try to send 5 seconds of audio in advance to initially fill Clementine's
// buffer.
g_object_set(G_OBJECT(tcpsink_), "ts-offset", qint64(-5 * kNsecPerSec),
nullptr);
// We know the time of each buffer
g_object_set(G_OBJECT(appsrc_), "format", GST_FORMAT_TIME, nullptr);
// Spotify only pushes data to us every 100ms, so keep the appsrc half full
// to prevent tiny stalls.
g_object_set(G_OBJECT(appsrc_), "min-percent", 50, nullptr);
// Set callbacks for when to start/stop pushing data
GstAppSrcCallbacks callbacks;
callbacks.enough_data = EnoughDataCallback;
callbacks.need_data = NeedDataCallback;
callbacks.seek_data = SeekDataCallback;
gst_app_src_set_callbacks(appsrc_, &callbacks, this, nullptr);
#if Q_BYTE_ORDER == Q_BIG_ENDIAN
static const char* format = "S16BE";
#elif Q_BYTE_ORDER == Q_LITTLE_ENDIAN
static const char* format = "S16LE";
#endif
// Set caps
GstCaps* caps = gst_caps_new_simple(
"audio/x-raw", "format", G_TYPE_STRING, format, "rate", G_TYPE_INT,
sample_rate, "channels", G_TYPE_INT, channels, "layout", G_TYPE_STRING,
"interleaved", nullptr);
gst_app_src_set_caps(appsrc_, caps);
gst_caps_unref(caps);
// Set size
byte_rate_ = quint64(sample_rate) * channels * 2;
const quint64 bytes = byte_rate_ * length_msec_ / 1000;
gst_app_src_set_size(appsrc_, bytes);
// Ready to go
return gst_element_set_state(pipeline_, GST_STATE_PLAYING) !=
GST_STATE_CHANGE_FAILURE;
}
void MediaPipeline::WriteData(const char* data, qint64 length) {
if (!is_initialised()) return;
GstBuffer* buffer = gst_buffer_new_allocate(nullptr, length, nullptr);
GstMapInfo map_info;
gst_buffer_map(buffer, &map_info, GST_MAP_WRITE);
memcpy(map_info.data, data, length);
gst_buffer_unmap(buffer, &map_info);
GST_BUFFER_PTS(buffer) = offset_bytes_ * kNsecPerSec / byte_rate_;
GST_BUFFER_DURATION(buffer) = length * kNsecPerSec / byte_rate_;
offset_bytes_ += length;
gst_app_src_push_buffer(appsrc_, buffer);
}
void MediaPipeline::EndStream() {
if (!is_initialised()) return;
gst_app_src_end_of_stream(appsrc_);
}
void MediaPipeline::NeedDataCallback(GstAppSrc* src, guint length, void* data) {
MediaPipeline* me = reinterpret_cast<MediaPipeline*>(data);
me->accepting_data_ = true;
}
void MediaPipeline::EnoughDataCallback(GstAppSrc* src, void* data) {
MediaPipeline* me = reinterpret_cast<MediaPipeline*>(data);
me->accepting_data_ = false;
}
gboolean MediaPipeline::SeekDataCallback(GstAppSrc* src, guint64 offset,
void* data) {
// MediaPipeline* me = reinterpret_cast<MediaPipeline*>(data);
qLog(Debug) << "Gstreamer wants seek to" << offset;
return false;
}